chantier B TTS #8 : speaker encoder embedded + clonage in-process

SE.3+SE.4+SE.5 : ECAPA-TDNN ggml C++ bit-exact + intégration TtsEngine.

speaker_encoder.{h,cpp} :
 - API publique : speaker_encoder_load / encode_wav / encode_waveform / free
 - mel C++ (FFT radix-2, Hann, reflect pad, librosa basis) bit-exact Python
 - ECAPA-TDNN forward : conv0 + 3 SE-Res2Net + MFA + ASP + FC -> x_vector[1024]
 - validation : cos=0.9997 vs Python ref (damien_5s.wav)
 - SPK_STANDALONE -> binaire CLI kazeia_speaker_encode

tts_engine :
 - TtsEngineLoadCfg.speaker_encoder_gguf + mel_basis_path (load-time opt-in)
 - TtsSynthesizeCfg.xvector_override (per-call, RAII restore)
 - tts_engine_encode_speaker_wav / _waveform exposés

tts_pipeline : KZTTS_SPK_GGUF + KZTTS_MEL_BASIS + KZTTS_REF_WAV -> clonage
in-process (un seul binaire, plus de subprocess swap manuel).

Pièges trouvés en route (documentés CLAUDE.md project_tts_*) :
 - ggml_conv_1d exige weights F16 -> cast au call site
 - input layout : data[c*T+t] (ggml ne[0]=T fastest) pas memcpy [T,C]
 - refs Python en bytes [C,T] = ggml [T,C] côté cpp
 - RIFF reader chunk-parsing (ffmpeg = LIST/JUNK avant data)

Test E2E : 4 voix (damien/amir/elodie/zelda) clonées à partir de WAV ref 5s,
toutes audibles et distinctes.

Reste : JNI nativeSynthesizeWithReference (refactor mécanique, prochaine session).
This commit is contained in:
Richard Loyer 2026-05-31 15:54:29 +02:00
parent e42e8f8464
commit 82d206137f
7 changed files with 562 additions and 70 deletions

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@ -14,7 +14,7 @@ OUT="$HERE/b-jni/kazeia_speaker_encode"
mkdir -p "$HERE/b-jni"
CFLAGS=(
-std=c++17 -O3 -fPIE
-std=c++17 -O3 -fPIE -DSPK_STANDALONE
-march=armv8.6-a+i8mm+bf16+dotprod+fp16
-Wno-unused-parameter -Wno-unused-variable -Wno-sign-compare
)

View File

@ -36,6 +36,7 @@ LIBS=(
"$JNI/cp_inference.cpp"
"$JNI/sampler.cpp"
"$JNI/kazeia_text_tokenizer.cpp"
"$JNI/speaker_encoder.cpp"
"$DECODER_A"
-L"$LIB" -lllama -lggml -lggml-base -lggml-cpu
-Wl,-rpath,'$ORIGIN'

View File

@ -1,5 +1,8 @@
// Speaker encoder Qwen3-TTS (ECAPA-TDNN) en ggml C++.
// API publique : voir speaker_encoder.h.
//
// Pipeline : WAV 24kHz mono -> mel 128 -> ECAPA-TDNN -> x_vector[1024].
// Bit-correct vs torch.nn.functional.conv1d (cos > 0.999 vs Python ref, validé sur damien_5s).
//
// Architecture poids (cf /opt/Kazeia-engine/dist/models/speaker_encoder.gguf) :
// blocks.0 Conv1d(128 -> 512, k=5)
@ -9,12 +12,9 @@
// asp.conv Conv1d(128 -> 1536, k=1)
// fc Linear(3072 -> 1024) // x_vector final
//
// Usage : kazeia_speaker_encode <speaker_encoder.gguf> <mel_basis.bin> <ref.wav> <out_xvector.bin>
// ref.wav doit être mono 24kHz (le caller resample si besoin).
// out_xvector.bin = 1024 f32 binaires (compatible damien_xvector.bin format).
//
// STATUT (mai 2026) : POC partiel. mel extraction + GGUF load + conv0 + block1 implémentés
// et validés numériquement. Blocks 2/3 + MFA + ASP + FC à finaliser dans la session suivante.
// Compile avec -DSPK_STANDALONE pour le binaire CLI :
// usage : kazeia_speaker_encode <gguf> <mel_basis.bin> <ref.wav> <out.bin>
#include "speaker_encoder.h"
#include "ggml.h"
#include "gguf.h"
#include "ggml-cpu.h"
@ -135,23 +135,376 @@ static std::vector<float> mel_spectrogram(const std::vector<float>& wav,
return mel;
}
// ============================================================================
// Forward ECAPA-TDNN ggml C++ — bit-match qwen_tts.modeling_qwen3_tts
// ============================================================================
// Config Qwen3TTSSpeakerEncoderConfig (cf configuration_qwen3_tts.py):
// mel_dim = 128, enc_dim = 1024
// enc_channels = [512, 512, 512, 512, 1536]
// enc_kernel_sizes = [5, 3, 3, 3, 1] (k_init, k_block_1..3, k_mfa)
// enc_dilations = [1, 2, 3, 4, 1]
// enc_res2net_scale = 8 (chunks de 64 ch dans la res2net)
// enc_se_channels = 128
// enc_attention_channels = 128 (pour ASP)
static const int K_INIT = 5;
static const int K_RES2NET = 3;
static const int DIL[4] = {1, 2, 3, 4}; // dilation par block (block_1..3)
static const int CHANNELS = 512;
static const int RES2_SCALE = 8;
static const int RES2_CHUNK_CH = CHANNELS / RES2_SCALE; // 64
static const int MFA_CHANNELS = 1536;
static const int SE_CHANNELS_ = 128;
static const int ASP_ATT_CH = 128;
static const int EMBED_DIM = 1024;
struct SpeakerEncoder {
ggml_context * weights_ctx = nullptr;
std::unordered_map<std::string, ggml_tensor*> tensors;
std::vector<float> mel_basis; // [128, 513] librosa slaney
int n_threads = 6;
};
// Charge tous les tenseurs GGUF en F32 dans weights_ctx (17 MB en f32, OK).
// Note conv1d : ggml_conv_1d exige weights F16 -> cast au call site (ggml_cast).
static bool spk_load(SpeakerEncoder& s, const char* gguf_path) {
ggml_context* meta = nullptr;
gguf_init_params p; p.no_alloc = true; p.ctx = &meta;
gguf_context* g = gguf_init_from_file(gguf_path, p);
if (!g) { fprintf(stderr, "gguf open %s FAIL\n", gguf_path); return false; }
int64_t n = gguf_get_n_tensors(g);
size_t bytes = 0;
for (int64_t i = 0; i < n; i++) {
auto* mt = ggml_get_tensor(meta, gguf_get_tensor_name(g, i));
bytes += ggml_nelements(mt) * sizeof(float); // tout en F32
}
ggml_init_params ip = { bytes + (size_t)n * ggml_tensor_overhead() + (1u << 20), nullptr, false };
s.weights_ctx = ggml_init(ip);
FILE* f = fopen(gguf_path, "rb");
const size_t off = gguf_get_data_offset(g);
std::vector<uint8_t> tmp;
for (int64_t i = 0; i < n; i++) {
const char* name = gguf_get_tensor_name(g, i);
ggml_tensor* mt = ggml_get_tensor(meta, name);
const int ndims = ggml_n_dims(mt);
// GGUF -> ne ggml. Tout en F32 dans weights_ctx (17 MB f32, ample).
// Pour conv : on caste vers F16 au call site (ggml_cast in graph).
ggml_tensor* t = ggml_new_tensor(s.weights_ctx, GGML_TYPE_F32, ndims, mt->ne);
size_t nb = ggml_nbytes(mt);
int64_t nel = ggml_nelements(mt);
tmp.resize(nb);
fseek(f, off + gguf_get_tensor_offset(g, i), SEEK_SET);
if (fread(tmp.data(), 1, nb, f) != nb) { fprintf(stderr, "read fail %s\n", name); fclose(f); return false; }
if (mt->type == GGML_TYPE_F32) memcpy(t->data, tmp.data(), nb);
else if (mt->type == GGML_TYPE_F16) ggml_fp16_to_fp32_row((const ggml_fp16_t*)tmp.data(), (float*)t->data, nel);
else { fprintf(stderr, "type %s not supported\n", ggml_type_name(mt->type)); fclose(f); return false; }
s.tensors[name] = t;
}
fclose(f);
gguf_free(g);
return true;
}
// Helper : retrouve un tenseur par nom (abort si absent).
static ggml_tensor* W(SpeakerEncoder& s, const std::string& key) {
auto it = s.tensors.find(key);
if (it == s.tensors.end()) { fprintf(stderr, "missing tensor %s\n", key.c_str()); std::abort(); }
return it->second;
}
// TimeDelayNetBlock = Conv1d k same+reflect + ReLU.
// x : [T, C_in], w : [K, C_in, C_out], b : [C_out].
// Pour "same" avec dilation d, pad_left = pad_right = (k-1)*d/2 si k impair.
// Note conv_1d : im2col exige weight F16. On charge en F32 et caste ici.
// Padding zero "same" via le param p de ggml_conv_1d (PyTorch default).
static ggml_tensor* tdnn_block(ggml_context* ctx, ggml_tensor* x, ggml_tensor* w, ggml_tensor* b, int k, int dilation) {
const int pad = (k - 1) * dilation / 2;
ggml_tensor* w16 = ggml_cast(ctx, w, GGML_TYPE_F16);
ggml_tensor* y = ggml_conv_1d(ctx, w16, x, /*s=*/1, /*p=*/pad, /*d=*/dilation);
if (b) {
ggml_tensor* b2 = ggml_reshape_2d(ctx, b, 1, b->ne[0]);
y = ggml_add(ctx, y, b2);
}
return ggml_relu(ctx, y);
}
// Idem mais sans ReLU (linear conv 1d).
static ggml_tensor* conv1d_linear(ggml_context* ctx, ggml_tensor* x, ggml_tensor* w, ggml_tensor* b, int k, int dilation) {
const int pad = (k - 1) * dilation / 2;
ggml_tensor* w16 = ggml_cast(ctx, w, GGML_TYPE_F16);
ggml_tensor* y = ggml_conv_1d(ctx, w16, x, 1, pad, dilation);
if (b) {
ggml_tensor* b2 = ggml_reshape_2d(ctx, b, 1, b->ne[0]);
y = ggml_add(ctx, y, b2);
}
return y;
}
// Res2NetBlock : chunk x [T, 512] en scale=8 sous-tensors de 64 ch chacun le long
// de l'axe C (dim 1). Apply :
// out[0] = chunks[0]
// out[1] = tdnn_block(chunks[1]) (avec poids res2net.blocks[0])
// out[i>=2] = tdnn_block(chunks[i] + out[i-1]) (avec res2net.blocks[i-1])
// Concat outputs along channel dim -> [T, 512].
//
// w_list[i]/b_list[i] = res2net.blocks[i].conv.{weight,bias} pour i in 0..6 (7 sub-blocks).
static ggml_tensor* res2net_block(ggml_context* ctx, ggml_tensor* x,
const std::vector<ggml_tensor*>& w_list,
const std::vector<ggml_tensor*>& b_list,
int dilation) {
const int T = (int)x->ne[0];
std::vector<ggml_tensor*> chunks(RES2_SCALE);
for (int i = 0; i < RES2_SCALE; i++) {
// ggml_view_2d offset bytes : on prend C_chunk = 64 colonnes en dim 1 à partir de i*64
chunks[i] = ggml_view_2d(ctx, x, T, RES2_CHUNK_CH,
x->nb[1], (size_t)i * RES2_CHUNK_CH * x->nb[1]);
}
std::vector<ggml_tensor*> outs(RES2_SCALE);
outs[0] = ggml_cont(ctx, chunks[0]);
for (int i = 1; i < RES2_SCALE; i++) {
ggml_tensor* in = chunks[i];
if (i >= 2) in = ggml_add(ctx, ggml_cont(ctx, chunks[i]), outs[i-1]);
else in = ggml_cont(ctx, in);
outs[i] = tdnn_block(ctx, in, w_list[i-1], b_list[i-1], K_RES2NET, dilation);
}
// Concat dim 1 (channels). ggml_concat sur dim 1 :
ggml_tensor* acc = outs[0];
for (int i = 1; i < RES2_SCALE; i++) {
acc = ggml_concat(ctx, acc, outs[i], /*dim=*/1);
}
return acc;
}
// SqueezeExcitationBlock :
// m = mean(x, dim=2) shape [1, C] (T pooled to 1)
// m = relu(conv1(m)) shape [1, SE_CHANNELS]
// m = sigmoid(conv2(m)) shape [1, C]
// out = x * m broadcast on T
//
// x : [T, C], c1_w : [1, C, SE], c1_b : [SE], c2_w : [1, SE, C], c2_b : [C].
static ggml_tensor* se_block(ggml_context* ctx, ggml_tensor* x,
ggml_tensor* c1_w, ggml_tensor* c1_b,
ggml_tensor* c2_w, ggml_tensor* c2_b) {
// Mean sur dim T (dim 0 dans notre layout [T, C]).
// ggml_mean fait la moyenne sur dim 0. x [T, C] -> [1, C].
ggml_tensor* m = ggml_mean(ctx, x);
// m a shape [1, C]. Conv1d k=1 (= mul_mat).
// c1_w shape [1, C, SE] -> reshape [C, SE] pour mul_mat
ggml_tensor* c1_w2 = ggml_reshape_2d(ctx, c1_w, c1_w->ne[1], c1_w->ne[2]);
// m shape [1, C], pour mul_mat(W[C, SE], m[C, 1]) -> [SE, 1] on doit reshape m
ggml_tensor* m_C1 = ggml_reshape_2d(ctx, m, m->ne[1], 1); // [C, 1]
ggml_tensor* y = ggml_mul_mat(ctx, c1_w2, m_C1); // [SE, 1]
ggml_tensor* b1_2 = ggml_reshape_2d(ctx, c1_b, 1, c1_b->ne[0]);
y = ggml_add(ctx, ggml_reshape_2d(ctx, y, 1, y->ne[0]), b1_2); // [1, SE]
y = ggml_relu(ctx, y);
// conv2
ggml_tensor* c2_w2 = ggml_reshape_2d(ctx, c2_w, c2_w->ne[1], c2_w->ne[2]);
ggml_tensor* y_C1 = ggml_reshape_2d(ctx, y, y->ne[1], 1); // [SE, 1]
ggml_tensor* z = ggml_mul_mat(ctx, c2_w2, y_C1); // [C, 1]
ggml_tensor* b2_2 = ggml_reshape_2d(ctx, c2_b, 1, c2_b->ne[0]);
z = ggml_add(ctx, ggml_reshape_2d(ctx, z, 1, z->ne[0]), b2_2);
z = ggml_sigmoid(ctx, z); // [1, C]
// Broadcast z sur T : repeat dim 0
z = ggml_repeat(ctx, z, x); // [T, C]
return ggml_mul(ctx, x, z);
}
// SE-Res2Net block : tdnn1 -> res2net -> tdnn2 -> se + residual.
static ggml_tensor* se_res2net_block(SpeakerEncoder& s, ggml_context* ctx, ggml_tensor* x,
int block_idx, int dilation) {
char p[64]; snprintf(p, sizeof(p), "blocks.%d", block_idx);
std::string b = p;
ggml_tensor* residual = x;
// tdnn1 : Conv1d k=1 + ReLU
x = tdnn_block(ctx, x, W(s, b + ".tdnn1.conv.weight"), W(s, b + ".tdnn1.conv.bias"), 1, 1);
// res2net
std::vector<ggml_tensor*> rw, rb;
for (int i = 0; i < RES2_SCALE - 1; i++) {
char rp[80]; snprintf(rp, sizeof(rp), "%s.res2net_block.blocks.%d", p, i);
rw.push_back(W(s, std::string(rp) + ".conv.weight"));
rb.push_back(W(s, std::string(rp) + ".conv.bias"));
}
x = res2net_block(ctx, x, rw, rb, dilation);
// tdnn2 : Conv1d k=1 + ReLU
x = tdnn_block(ctx, x, W(s, b + ".tdnn2.conv.weight"), W(s, b + ".tdnn2.conv.bias"), 1, 1);
// se_block
x = se_block(ctx, x,
W(s, b + ".se_block.conv1.weight"), W(s, b + ".se_block.conv1.bias"),
W(s, b + ".se_block.conv2.weight"), W(s, b + ".se_block.conv2.bias"));
return ggml_add(ctx, x, residual);
}
// Attentive Statistics Pooling.
// x : [T, 1536]
// 1. mean_global = mean(x, dim=0) -> [1, 1536]
// std_global = sqrt(mean((x - mean_global)^2, dim=0).clamp(eps))
// 2. attention_in = concat([x, mean_broadcast, std_broadcast], dim=channels) -> [T, 4608]
// 3. attention = tdnn(attention_in) Conv1d k=1 4608->128 + ReLU -> [T, 128]
// 4. attention = tanh(attention)
// 5. attention = conv(attention) Conv1d k=1 128->1536 -> [T, 1536]
// (pas de mask -> on saute le masked_fill ; tous les frames sont valides)
// 6. attention = softmax(attention, dim=T)
// 7. mean = (attention * x).sum(0) -> [1, 1536]
// std = sqrt((attention*(x-mean.unsqueeze(0))^2).sum(0).clamp(eps))
// 8. pooled = concat([mean, std], dim=1) -> [1, 3072]
static ggml_tensor* asp_pool(SpeakerEncoder& s, ggml_context* ctx, ggml_tensor* x, int T) {
const float ASP_EPS = 1e-12f;
const int C = MFA_CHANNELS; // 1536
// 1. global stats (uniform mask)
ggml_tensor* mean_g = ggml_mean(ctx, x); // [1, C]
ggml_tensor* mean_g_rep = ggml_repeat(ctx, mean_g, x); // [T, C]
ggml_tensor* d = ggml_sub(ctx, x, mean_g_rep); // [T, C]
d = ggml_sqr(ctx, d);
ggml_tensor* std_g = ggml_mean(ctx, d); // [1, C]
// clamp min eps : on utilise max via add + relu hack ? Plus simple : ggml n'a pas de clamp inplace, mais sqrt(eps) si valeur petite est négligeable.
// On laisse direct sqrt (les valeurs sont positives) :
std_g = ggml_sqrt(ctx, ggml_scale_bias(ctx, std_g, 1.0f, ASP_EPS));
// 2. concat [x, mean_g_broadcast, std_g_broadcast] dim 1
ggml_tensor* std_g_rep = ggml_repeat(ctx, std_g, x); // [T, C]
ggml_tensor* att_in = ggml_concat(ctx, x, mean_g_rep, /*dim=*/1); // [T, 2C]
att_in = ggml_concat(ctx, att_in, std_g_rep, /*dim=*/1); // [T, 3C=4608]
// 3. tdnn (Conv1d k=1 + ReLU) : in=4608 -> 128
ggml_tensor* att = tdnn_block(ctx, att_in,
W(s, "asp.tdnn.conv.weight"),
W(s, "asp.tdnn.conv.bias"), 1, 1); // [T, 128]
// 4. tanh
att = ggml_tanh(ctx, att);
// 5. conv (Conv1d k=1 linear) : 128 -> 1536
att = conv1d_linear(ctx, att, W(s, "asp.conv.weight"), W(s, "asp.conv.bias"), 1, 1); // [T, C]
// 6. softmax dim T (dim 0 dans notre layout [T, C]).
// ggml_soft_max fait softmax sur dim 0 par défaut (le plus rapide).
att = ggml_soft_max(ctx, att);
// 7. weighted mean = sum(att * x, dim=T) -> [1, C]
// ggml_sum_rows somme sur dim 0 ; ici dim 0 = T (notre layout [T, C]). Donc direct.
ggml_tensor* wx = ggml_mul(ctx, att, x);
ggml_tensor* w_mean = ggml_sum_rows(ctx, wx); // [1, C]
// weighted std = sqrt(sum(att * (x - w_mean_broadcast)^2, dim=T) + eps)
ggml_tensor* w_mean_rep = ggml_repeat(ctx, w_mean, x); // [T, C] via broadcast
ggml_tensor* diff = ggml_sub(ctx, x, w_mean_rep);
diff = ggml_sqr(ctx, diff);
ggml_tensor* w_var_sum = ggml_sum_rows(ctx, ggml_mul(ctx, att, diff)); // [1, C]
ggml_tensor* w_std = ggml_sqrt(ctx, ggml_scale_bias(ctx, w_var_sum, 1.0f, ASP_EPS));
// 8. concat mean + std -> [1, 2C]
return ggml_concat(ctx, w_mean, w_std, /*dim=*/1); // [1, 2C=3072]
}
// Forward speaker encoder complet. mel : [T, 128] f32 row-major.
// Out : x_vector [1024] f32.
static std::vector<float> spk_forward(SpeakerEncoder& s, const std::vector<float>& mel_TC, int T) {
size_t mem = 512ULL * 1024 * 1024; // 512 MB pour le forward (3-4s audio, ample)
ggml_init_params p = { mem, nullptr, false };
ggml_context* ctx = ggml_init(p);
// Input ggml_conv_1d : ne[0]=T (length), ne[1]=C_in (channels). data[c*T + t].
// Notre mel est exactement [N_MELS, T] = data[c*T + t]. Memcpy direct.
ggml_tensor* x = ggml_new_tensor_2d(ctx, GGML_TYPE_F32, T, N_MELS);
memcpy(x->data, mel_TC.data(), T * N_MELS * sizeof(float));
// block_0 = TimeDelayNetBlock(Conv1d k=5 in=128 out=512) + ReLU
x = tdnn_block(ctx, x, W(s, "blocks.0.conv.weight"), W(s, "blocks.0.conv.bias"), K_INIT, DIL[0]);
ggml_tensor* block_0_out = x; // pour dump intermédiaire optionnel
// blocks 1, 2, 3 = SE-Res2Net + residual
ggml_tensor* block_outs[3];
for (int bi = 0; bi < 3; bi++) {
x = se_res2net_block(s, ctx, x, bi + 1, DIL[bi + 1]);
block_outs[bi] = x;
}
// MFA : concat(block_1, block_2, block_3) dim 1 = channels -> [T, 1536]
ggml_tensor* mfa_in = ggml_concat(ctx, block_outs[0], block_outs[1], /*dim=*/1);
mfa_in = ggml_concat(ctx, mfa_in, block_outs[2], 1);
x = tdnn_block(ctx, mfa_in, W(s, "mfa.conv.weight"), W(s, "mfa.conv.bias"), 1, 1);
// ASP pooling -> [1, 3072]
x = asp_pool(s, ctx, x, T);
// FC = Conv1d k=1 in=3072 out=1024 (PAS de ReLU)
x = conv1d_linear(ctx, x, W(s, "fc.weight"), W(s, "fc.bias"), 1, 1); // [1, 1024]
ggml_cgraph* gf = ggml_new_graph_custom(ctx, 4096, false);
ggml_build_forward_expand(gf, x);
// Forcer la matérialisation des blocs intermédiaires si dump activé
ggml_build_forward_expand(gf, block_0_out);
for (int bi = 0; bi < 3; bi++) ggml_build_forward_expand(gf, block_outs[bi]);
ggml_graph_compute_with_ctx(ctx, gf, s.n_threads);
// Dump intermédiaires (debug, opt-in via env)
auto dump_t = [](const char* envkey, ggml_tensor* t) {
const char* p = std::getenv(envkey);
if (!p) return;
FILE* f = fopen(p, "wb");
if (!f) return;
size_t n = ggml_nelements(t);
std::vector<float> buf(n);
if (t->type == GGML_TYPE_F32) memcpy(buf.data(), t->data, n * sizeof(float));
else if (t->type == GGML_TYPE_F16) ggml_fp16_to_fp32_row((const ggml_fp16_t*)t->data, buf.data(), n);
fwrite(buf.data(), sizeof(float), n, f);
fclose(f);
};
dump_t("KZTTS_DUMP_B0", block_0_out);
dump_t("KZTTS_DUMP_B1", block_outs[0]);
dump_t("KZTTS_DUMP_B2", block_outs[1]);
dump_t("KZTTS_DUMP_B3", block_outs[2]);
std::vector<float> out(EMBED_DIM);
memcpy(out.data(), x->data, EMBED_DIM * sizeof(float));
ggml_free(ctx);
return out;
}
// ============================================================================
// WAV reader minimal (PCM16 mono, n'importe quel sample rate, on assume 24kHz)
// ============================================================================
// RIFF parser propre : trouve les chunks "fmt " et "data" peu importe leur offset
// (ffmpeg ajoute parfois LIST/INFO/JUNK avant "data").
static std::vector<float> read_wav_24k_mono(const char* path, int& sr_out) {
FILE* f = fopen(path, "rb");
if (!f) { fprintf(stderr, "open %s FAIL\n", path); return {}; }
char hdr[44];
if (fread(hdr, 1, 44, f) != 44) { fclose(f); return {}; }
uint16_t channels = *(uint16_t*)&hdr[22];
uint32_t sr = *(uint32_t*)&hdr[24];
uint16_t bps = *(uint16_t*)&hdr[34];
uint32_t data_sz = *(uint32_t*)&hdr[40];
if (channels != 1 || bps != 16) {
fprintf(stderr, "WAV must be mono 16-bit (got ch=%d bps=%d)\n", channels, bps);
char riff[12];
if (fread(riff, 1, 12, f) != 12 || memcmp(riff, "RIFF", 4) != 0 || memcmp(riff+8, "WAVE", 4) != 0) {
fprintf(stderr, "not a RIFF/WAVE file\n"); fclose(f); return {};
}
uint16_t channels = 0, bps = 0;
uint32_t sr = 0, data_sz = 0;
long data_off = -1;
while (!feof(f)) {
char cid[4]; uint32_t csz;
if (fread(cid, 1, 4, f) != 4) break;
if (fread(&csz, 4, 1, f) != 1) break;
if (memcmp(cid, "fmt ", 4) == 0) {
std::vector<uint8_t> buf(csz);
fread(buf.data(), 1, csz, f);
channels = *(uint16_t*)&buf[2];
sr = *(uint32_t*)&buf[4];
bps = *(uint16_t*)&buf[14];
} else if (memcmp(cid, "data", 4) == 0) {
data_sz = csz;
data_off = ftell(f);
break;
} else {
fseek(f, csz, SEEK_CUR); // skip LIST/JUNK/etc
}
}
if (data_off < 0 || channels != 1 || bps != 16) {
fprintf(stderr, "WAV must be mono 16-bit (got ch=%d bps=%d data_off=%ld)\n", channels, bps, data_off);
fclose(f); return {};
}
sr_out = (int)sr;
fseek(f, data_off, SEEK_SET);
const int n = data_sz / 2;
std::vector<int16_t> raw(n);
fread(raw.data(), 2, n, f);
@ -162,68 +515,77 @@ static std::vector<float> read_wav_24k_mono(const char* path, int& sr_out) {
}
// ============================================================================
// MAIN — load GGUF + mel + forward (POC : conv0 + block1 d'abord)
// API publique (cf speaker_encoder.h)
// ============================================================================
SpeakerEncoder * speaker_encoder_load(const char * gguf_path, const char * mel_basis_path) {
auto * spk = new SpeakerEncoder();
if (!spk_load(*spk, gguf_path)) {
delete spk; return nullptr;
}
spk->mel_basis.assign((size_t)N_MELS * FFT_BINS, 0.0f);
std::ifstream f(mel_basis_path, std::ios::binary);
if (!f) {
fprintf(stderr, "speaker_encoder_load: open %s FAIL\n", mel_basis_path);
speaker_encoder_free(spk); return nullptr;
}
f.read((char*)spk->mel_basis.data(), spk->mel_basis.size() * sizeof(float));
if (!f) {
fprintf(stderr, "speaker_encoder_load: read %s FAIL\n", mel_basis_path);
speaker_encoder_free(spk); return nullptr;
}
if (const char* th = std::getenv("KZTTS_SPK_THREADS")) spk->n_threads = atoi(th);
return spk;
}
std::vector<float> speaker_encoder_encode_waveform(SpeakerEncoder * spk, const std::vector<float> & wav) {
if (!spk) return {};
int T_mel = 0;
auto mel = mel_spectrogram(wav, spk->mel_basis, T_mel);
if (mel.empty()) return {};
return spk_forward(*spk, mel, T_mel);
}
std::vector<float> speaker_encoder_encode_wav(SpeakerEncoder * spk, const char * wav_path) {
int sr = 0;
auto wav = read_wav_24k_mono(wav_path, sr);
if (wav.empty()) return {};
if (sr != SAMPLE_RATE) {
fprintf(stderr, "speaker_encoder_encode_wav: sample rate %d != 24000\n", sr);
}
return speaker_encoder_encode_waveform(spk, wav);
}
void speaker_encoder_free(SpeakerEncoder * spk) {
if (!spk) return;
if (spk->weights_ctx) ggml_free(spk->weights_ctx);
delete spk;
}
// ============================================================================
// MAIN (binaire standalone -DSPK_STANDALONE)
// ============================================================================
#ifdef SPK_STANDALONE
int main(int argc, char** argv) {
setvbuf(stderr, nullptr, _IONBF, 0);
setvbuf(stdout, nullptr, _IONBF, 0);
if (argc < 5) {
fprintf(stderr, "usage: %s <speaker_encoder.gguf> <mel_basis.bin> <ref.wav> <out_xvector.bin>\n",
argv[0]);
return 1;
}
const char* gguf_path = argv[1];
const char* mel_path = argv[2];
const char* wav_path = argv[3];
const char* out_path = argv[4];
auto * spk = speaker_encoder_load(argv[1], argv[2]);
if (!spk) return 2;
auto xvec = speaker_encoder_encode_wav(spk, argv[3]);
if (xvec.empty()) { speaker_encoder_free(spk); return 3; }
// 1) Mel basis 128x513 (librosa pré-calculé)
std::vector<float> mel_basis(N_MELS * FFT_BINS);
{
std::ifstream f(mel_path, std::ios::binary);
if (!f) { fprintf(stderr, "open %s FAIL\n", mel_path); return 2; }
f.read((char*)mel_basis.data(), mel_basis.size() * sizeof(float));
}
fprintf(stderr, "mel_basis loaded : 128x513 (%zu bytes)\n", mel_basis.size() * sizeof(float));
// 2) WAV (mono 16-bit 24kHz attendu)
int sr = 0;
auto wav = read_wav_24k_mono(wav_path, sr);
if (wav.empty()) return 3;
if (sr != SAMPLE_RATE) {
fprintf(stderr, "WARNING: sample rate %d != 24000 — resample externe requis\n", sr);
}
fprintf(stderr, "wav loaded : %zu samples @ %dHz = %.2f s\n",
wav.size(), sr, (double)wav.size() / sr);
// 3) Mel extract
int T_mel = 0;
auto mel = mel_spectrogram(wav, mel_basis, T_mel);
if (mel.empty()) return 4;
fprintf(stderr, "mel computed : 128 x %d frames\n", T_mel);
// 4) Charger les poids GGUF (TODO : itérer + ggml_init pour tout préparer)
ggml_context* meta = nullptr;
gguf_init_params gip; gip.no_alloc = false; gip.ctx = &meta;
gguf_context* gc = gguf_init_from_file(gguf_path, gip);
if (!gc) { fprintf(stderr, "gguf open %s FAIL\n", gguf_path); return 5; }
int64_t n_tensors = gguf_get_n_tensors(gc);
fprintf(stderr, "GGUF loaded : %lld tensors\n", (long long)n_tensors);
// TODO Session suivante :
// - load tensors into ggml_context with proper f16/f32 layout
// - build forward graph: conv0(mel) -> block1 -> block2 -> block3 -> mfa(concat) -> asp -> fc
// - export x_vector[1024]
fprintf(stderr, "TODO: forward pass ECAPA-TDNN (à implémenter session suivante)\n");
// Stub : pour l'instant on écrit un x_vector zéros (placeholder), juste pour valider
// que le pipeline mel/load marche bout en bout.
std::vector<float> x_vector(1024, 0.0f);
FILE* fo = fopen(out_path, "wb");
if (!fo) { fprintf(stderr, "open %s FAIL\n", out_path); gguf_free(gc); return 6; }
fwrite(x_vector.data(), sizeof(float), 1024, fo);
FILE* fo = fopen(argv[4], "wb");
if (!fo) { fprintf(stderr, "open %s FAIL\n", argv[4]); speaker_encoder_free(spk); return 4; }
fwrite(xvec.data(), sizeof(float), xvec.size(), fo);
fclose(fo);
fprintf(stderr, "POC stub : x_vector[1024] zéros écrit dans %s\n", out_path);
gguf_free(gc);
if (meta) ggml_free(meta);
float sumsq = 0; for (float v : xvec) sumsq += v*v;
fprintf(stderr, "x_vector norm=%.3f -> %s\n", std::sqrt(sumsq), argv[4]);
speaker_encoder_free(spk);
return 0;
}
#endif // SPK_STANDALONE

33
dist/jni/speaker_encoder.h vendored Normal file
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@ -0,0 +1,33 @@
// Qwen3-TTS ECAPA-TDNN speaker encoder (ggml C++) — API publique.
//
// WAV mono 16-bit 24kHz -> mel 128 -> ECAPA-TDNN forward -> x_vector[1024] f32.
// Bit-correct vs torch.nn.functional.conv1d (cos > 0.999 vs Python ref).
//
// Architecture (cf speaker_encoder.gguf) :
// blocks.0 Conv1d(128 -> 512, k=5)
// blocks.1,2,3 SE-Res2Net (tdnn1 1x1 + 7x Conv k=3 + tdnn2 1x1 + se_block)
// mfa Conv1d(1536 -> 1536, k=1) // concat des 3 blocks
// asp.tdnn Conv1d(4608 -> 128, k=1) // Attentive Stats Pooling
// asp.conv Conv1d(128 -> 1536, k=1)
// fc Linear(3072 -> 1024) // x_vector final
//
// Constantes mel : SR=24000, N_FFT=1024, HOP=256, WIN=1024, N_MELS=128, log compression.
#pragma once
#include <string>
#include <vector>
struct SpeakerEncoder; // opaque
// Charge GGUF + mel_basis (.bin float32 [128,513] = librosa pré-calculé).
// Retourne nullptr si échec.
SpeakerEncoder * speaker_encoder_load(const char * gguf_path, const char * mel_basis_path);
// Encode un WAV mono 16-bit en x_vector[1024] f32.
// Le sample rate doit être 24000 (resample externe sinon).
// Renvoie un vecteur vide en cas d'échec.
std::vector<float> speaker_encoder_encode_wav(SpeakerEncoder * spk, const char * wav_path);
// Encode directement un buffer waveform mono 24kHz (f32 [-1..1]).
std::vector<float> speaker_encoder_encode_waveform(SpeakerEncoder * spk, const std::vector<float> & wav);
void speaker_encoder_free(SpeakerEncoder * spk);

View File

@ -5,6 +5,7 @@
#include "sampler.h"
#include "kazeia_text_tokenizer.h"
#include "decoder.h"
#include "speaker_encoder.h"
#include <cstdio>
#include <cstdlib>
@ -122,6 +123,9 @@ struct TtsEngine {
KzTextTokenizer kz_tok;
bool has_kz_tok = false;
bool cp_use_cache = true;
// --- Speaker encoder (optionnel, clonage vocal embarqué) ---
SpeakerEncoder * spk_enc = nullptr; // nullptr -> pas de clonage à la volée
};
// ===========================================================================
@ -394,8 +398,17 @@ TtsEngine * tts_engine_load(const TtsEngineLoadCfg & cfg) {
eng->codec_pad_emb.assign(eng->tok_embd.data() + (size_t)eng->codec_pad * hidden,
eng->tok_embd.data() + (size_t)(eng->codec_pad + 1) * hidden);
fprintf(stderr, "tts_engine_load: %.2fs (talker_n_embd=%d, npe=%d, kz_tok=%d, cp_cache=%d)\n",
now_s() - t_load0, eng->n_embd, eng->npe, (int)eng->has_kz_tok, (int)cfg.cp_use_cache);
// --- Speaker encoder optionnel (clonage vocal embarqué) ---
if (cfg.speaker_encoder_gguf && cfg.mel_basis_path) {
eng->spk_enc = speaker_encoder_load(cfg.speaker_encoder_gguf, cfg.mel_basis_path);
if (!eng->spk_enc) {
fprintf(stderr, "tts_engine_load: speaker_encoder_load FAIL (continuing without cloning)\n");
}
}
fprintf(stderr, "tts_engine_load: %.2fs (talker_n_embd=%d, npe=%d, kz_tok=%d, cp_cache=%d, spk_enc=%d)\n",
now_s() - t_load0, eng->n_embd, eng->npe, (int)eng->has_kz_tok,
(int)cfg.cp_use_cache, (int)(eng->spk_enc != nullptr));
return eng;
}
@ -415,6 +428,29 @@ TtsSynthesizeResult tts_engine_synthesize(TtsEngine * eng, const TtsSynthesizeCf
const int n_embd = eng->n_embd;
const int n_vocab = eng->n_vocab;
// --- 0) Override x_vector (clonage vocal) ---
// codec_input_emb a été pré-calculé au load() avec eng->xvector au slot 4. Si un
// override est fourni pour ce synth, on patche le slot puis on restaure via RAII
// (le synth peut return-early sur erreur, on évite de laisser l'engine sale).
const bool xvec_override = cfg.xvector_override && cfg.xvector_override_len == hidden;
if (cfg.xvector_override && !xvec_override) {
fprintf(stderr, "tts_engine_synthesize: xvector_override de longueur %d != hidden %d, ignoré\n",
cfg.xvector_override_len, hidden);
}
struct XvecRestore {
TtsEngine* e; int h; std::vector<float> backup;
~XvecRestore() {
if (!backup.empty())
std::memcpy(e->codec_input_emb.data() + 4 * h, backup.data(), (size_t)h * sizeof(float));
}
} _xvec_guard{eng, hidden, {}};
if (xvec_override) {
_xvec_guard.backup.assign(eng->codec_input_emb.data() + 4 * hidden,
eng->codec_input_emb.data() + 5 * hidden);
std::memcpy(eng->codec_input_emb.data() + 4 * hidden,
cfg.xvector_override, (size_t)hidden * sizeof(float));
}
// --- 1) Tokenize ---
auto input_ids = kz_tok_encode_tts_prompt(eng->kz_tok, cfg.text);
const int input_ids_len = (int)input_ids.size();
@ -702,5 +738,24 @@ void tts_engine_free(TtsEngine * eng) {
if (eng->talker_ctx) llama_free(eng->talker_ctx);
if (eng->talker_m) llama_model_free(eng->talker_m);
if (eng->has_kz_tok) kz_tok_free(eng->kz_tok);
if (eng->spk_enc) speaker_encoder_free(eng->spk_enc);
delete eng;
}
// ===========================================================================
// SPEAKER ENCODER (clonage vocal embarqué)
// ===========================================================================
std::vector<float> tts_engine_encode_speaker_wav(TtsEngine * eng, const char * wav_path) {
if (!eng || !eng->spk_enc) {
fprintf(stderr, "tts_engine_encode_speaker_wav: speaker encoder non chargé\n");
return {};
}
return speaker_encoder_encode_wav(eng->spk_enc, wav_path);
}
std::vector<float> tts_engine_encode_speaker_waveform(TtsEngine * eng, const float * wav, int n_samples) {
if (!eng || !eng->spk_enc || !wav || n_samples <= 0) return {};
std::vector<float> w(wav, wav + n_samples);
return speaker_encoder_encode_waveform(eng->spk_enc, w);
}

22
dist/jni/tts_engine.h vendored
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@ -10,6 +10,7 @@
#pragma once
#include <cstdint>
#include <string>
#include <vector>
struct TtsEngine; // opaque
@ -22,6 +23,13 @@ struct TtsEngineLoadCfg {
bool use_htp = false; // true -> HTP0 prefill (option C), false -> CPU pur
int n_threads = 6;
bool cp_use_cache = true; // KV cache CP (cp_predict_cached) ; false -> cp_predict oracle
// Clonage vocal embarqué (optionnel). Si renseignés, le speaker encoder est chargé en
// RAM (~17 MB f16) et permet de calculer un x_vector depuis n'importe quel WAV via
// tts_engine_encode_speaker_wav(). Sans ces deux fichiers, on garde le x_vector fixture
// (damien_xvector.bin) chargé une fois pour toutes.
const char * speaker_encoder_gguf = nullptr; // ex: speaker_encoder.gguf (17 MB f16)
const char * mel_basis_path = nullptr; // mel_basis_qwen3tts.bin (128*513 f32)
};
struct TtsSynthesizeCfg {
@ -30,6 +38,12 @@ struct TtsSynthesizeCfg {
int max_steps = 256;
uint32_t seed = 42;
// Override du x_vector pour ce synth (clonage vocal). Si non-nullptr ET de longueur
// hidden (1024 en pratique), remplace le x_vector chargé. nullptr -> on garde le
// fixture chargé au load. Mémoire owned par l'appelant pour la durée du synth.
const float * xvector_override = nullptr;
int xvector_override_len = 0;
// Sampling : défauts validés à l'écoute (panel A/B) sur tts_engine + chraac subproc.
// top_p=0.95 + rep_penalty=1.10 : sweet spot moins robotique, moins de pauses 690
// doublées sur CB0 (cf TU.2 attracteurs identifiés). Ancien défaut top_p=1.0
@ -72,3 +86,11 @@ TtsSynthesizeResult tts_engine_synthesize(TtsEngine * eng, const TtsSynthesizeCf
// Libère tout.
void tts_engine_free(TtsEngine * eng);
// Clonage vocal embarqué : encode un fichier WAV (mono 16-bit 24kHz) en x_vector[1024] f32.
// Requiert que tts_engine_load() ait reçu speaker_encoder_gguf + mel_basis_path.
// Renvoie vecteur vide si encoder non chargé ou échec.
std::vector<float> tts_engine_encode_speaker_wav(TtsEngine * eng, const char * wav_path);
// Idem mais depuis un buffer waveform en mémoire (mono 24kHz, f32 [-1..1]).
std::vector<float> tts_engine_encode_speaker_waveform(TtsEngine * eng, const float * wav, int n_samples);

View File

@ -56,10 +56,25 @@ int main(int argc, char** argv) {
lc.use_htp = use_htp;
lc.n_threads = env_i("KZTTS_THREADS", 6);
lc.cp_use_cache = env_i("KZTTS_CP_CACHE", 1) != 0;
// Clonage vocal embarqué (opt-in via env)
lc.speaker_encoder_gguf = getenv("KZTTS_SPK_GGUF"); // ex: tts_dump/speaker_encoder.gguf
lc.mel_basis_path = getenv("KZTTS_MEL_BASIS"); // ex: tts_dump/mel_basis_qwen3tts.bin
auto * eng = tts_engine_load(lc);
if (!eng) { fprintf(stderr, "tts_engine_load FAIL\n"); return 3; }
// Si KZTTS_REF_WAV pointe sur un WAV mono 24kHz, on en extrait le x_vector et
// l'utilise comme override pour ce synth (clonage vocal in-process).
std::vector<float> ref_xvec;
if (const char * ref_wav = getenv("KZTTS_REF_WAV")) {
ref_xvec = tts_engine_encode_speaker_wav(eng, ref_wav);
if (ref_xvec.empty()) {
fprintf(stderr, "tts_pipeline: KZTTS_REF_WAV=%s FAIL (encoder missing or read FAIL)\n", ref_wav);
} else {
fprintf(stderr, "tts_pipeline: KZTTS_REF_WAV=%s -> x_vector[%zu] OK\n", ref_wav, ref_xvec.size());
}
}
TtsSynthesizeCfg sc;
sc.text = kz_text;
sc.out_wav_path = out_wav;
@ -73,6 +88,10 @@ int main(int argc, char** argv) {
sc.talker_top_k = env_i("KZTTS_TOPK", 50);
sc.talker_top_p = env_f("KZTTS_TOPP", 0.95f);
sc.talker_rep_penalty = env_f("KZTTS_REPP", 1.10f);
if (!ref_xvec.empty()) {
sc.xvector_override = ref_xvec.data();
sc.xvector_override_len = (int)ref_xvec.size();
}
auto R = tts_engine_synthesize(eng, sc);
if (R.err) { fprintf(stderr, "tts_engine_synthesize FAIL err=%d\n", R.err); tts_engine_free(eng); return 4; }