Files
llama-cpp/tools/mtmd/mtmd-audio.h
T
Pascal ba3c05f71d
Copilot Setup Steps / copilot-setup-steps (push) Has been cancelled
Check Pre-Tokenizer Hashes / pre-tokenizer-hashes (push) Has been cancelled
Python check requirements.txt / check-requirements (push) Has been cancelled
Python Type-Check / python type-check (push) Has been cancelled
Update Operations Documentation / update-ops-docs (push) Has been cancelled
model: add Muse Spark Open (text, vision and audio) (#30616)
* model: add Muse Spark Open (text, vision and audio)

Muse Spark Open is a latent space MoE: the residual stream stays at 8192
while the 256 experts live in 4096, with a projection down and up around
them, top-8 sigmoid routing with a correction bias. Attention interleaves
three sliding layers at a 2048 window with rope for one global layer
without rope, with a sigmoid output gate and a weightless QK-norm.
Residuals are norm preserving, their gate is folded at conversion time.

Vision reuses the Muse Glimmer tower. Audio is a causal convolutional
pyramid with no attention and no spectrogram, four stages folding 16, 6,
5 and 4 samples into the channel dim, one embedding per 80 ms of 24 kHz.

The quantizer keeps the latent projections and the attention gate at
Q8_0, a little over 1% of the weights, since the whole residual stream
passes through them and the low-bit types turned the output into a
repeating loop. The model saver writes the post norm epsilon, and the
arch is registered in test-llama-archs.

Co-authored-by: Xuan Son Nguyen <son@huggingface.co>
Co-authored-by: Pedro Cuenca <pedro@huggingface.co>

* Muse Spark: drop the redundant forward expand

* Muse Spark: drop the redundant forward expand and tighten the comments

* Update src/llama-quant.cpp

Co-authored-by: Sigbjørn Skjæret <sigbjorn.skjaeret@huggingface.co>

* Update src/llama-graph.cpp

Co-authored-by: Sigbjørn Skjæret <sigbjorn.skjaeret@huggingface.co>

* Update conversion/muse_spark.py

Co-authored-by: Sigbjørn Skjæret <sigbjorn.skjaeret@huggingface.co>

* Update conversion/muse_spark.py

Co-authored-by: Sigbjørn Skjæret <sigbjorn.skjaeret@huggingface.co>

* Update conversion/muse_spark.py

Co-authored-by: Sigbjørn Skjæret <sigbjorn.skjaeret@huggingface.co>

* Update conversion/muse_spark.py

Co-authored-by: Sigbjørn Skjæret <sigbjorn.skjaeret@huggingface.co>

---------

Co-authored-by: Xuan Son Nguyen <son@huggingface.co>
Co-authored-by: Pedro Cuenca <pedro@huggingface.co>
Co-authored-by: Sigbjørn Skjæret <sigbjorn.skjaeret@huggingface.co>
2026-10-15 23:27:12 +02:00

211 lines
7.7 KiB
C++
Raw Blame History

This file contains ambiguous Unicode characters
This file contains Unicode characters that might be confused with other characters. If you think that this is intentional, you can safely ignore this warning. Use the Escape button to reveal them.
#pragma once
#include "ggml.h"
#include "clip-model.h"
#include <cstdint>
#include <vector>
#include <string>
#define MTMD_INTERNAL_HEADER
struct mtmd_audio_mel {
int64_t n_len;
int64_t n_len_org;
int64_t n_mel;
std::vector<float> data;
};
struct mtmd_audio_mel_filters {
int64_t n_mel;
int64_t n_fft;
std::vector<float> data;
};
// cache for audio processing, each processor instance owns its own cache
struct mtmd_audio_cache {
std::vector<float> sin_vals;
std::vector<float> cos_vals;
std::vector<float> hann_window;
mtmd_audio_mel_filters filters;
void fill_sin_cos_table(uint32_t n);
void fill_hann_window(uint32_t length, bool periodic);
// Build mel filterbank matrix [n_mel × n_fft_bins] at runtime.
// n_fft_bins must be (N_fft / 2 + 1). Example: if N_fft=512 -> n_fft_bins=257.
void fill_mel_filterbank_matrix(int64_t n_mel,
int64_t n_fft,
int sample_rate, // e.g. 16000
float fmin = 0.0f, // e.g. 0.0
float fmax = -1.0f, // e.g. sr/2; pass -1 for auto
bool slaney_area_norm = true,
float scale = 1.0f,
bool use_htk = false
);
};
struct mtmd_audio_preprocessor {
const clip_hparams & hparams;
mtmd_audio_preprocessor(const clip_ctx * ctx): hparams(*clip_get_hparams(ctx)) {}
mtmd_audio_preprocessor(const clip_hparams & hparams): hparams(hparams) {}
virtual ~mtmd_audio_preprocessor() = default;
virtual void initialize() = 0; // NOT thread-safe
virtual bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) const = 0;
};
struct mtmd_audio_preprocessor_whisper : mtmd_audio_preprocessor {
mtmd_audio_preprocessor_whisper(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
void initialize() override;
bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) const override;
private:
mtmd_audio_cache cache;
};
struct mtmd_audio_preprocessor_muse_spark_audio : mtmd_audio_preprocessor {
mtmd_audio_preprocessor_muse_spark_audio(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
void initialize() override {}
bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) const override;
};
struct mtmd_audio_preprocessor_conformer : mtmd_audio_preprocessor {
mtmd_audio_preprocessor_conformer(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
mtmd_audio_preprocessor_conformer(const clip_hparams & hparams) : mtmd_audio_preprocessor(hparams) {}
void initialize() override;
bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) const override;
private:
mtmd_audio_cache cache;
};
// same as conformer, the audio is cut to 30 s and padded to 0.5 s (d1-omni audio.py)
struct mtmd_audio_preprocessor_d1omni : mtmd_audio_preprocessor_conformer {
using mtmd_audio_preprocessor_conformer::mtmd_audio_preprocessor_conformer;
bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) const override;
};
struct mtmd_audio_preprocessor_granite_speech : mtmd_audio_preprocessor {
mtmd_audio_preprocessor_granite_speech(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
void initialize() override;
bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) const override;
private:
mtmd_audio_cache cache;
};
struct mtmd_audio_preprocessor_gemma4a : mtmd_audio_preprocessor {
mtmd_audio_preprocessor_gemma4a(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
void initialize() override;
bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) const override;
private:
mtmd_audio_cache cache;
};
struct mtmd_audio_preprocessor_gemma4ua : mtmd_audio_preprocessor {
mtmd_audio_preprocessor_gemma4ua(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
void initialize() override;
bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) const override;
};
struct mtmd_audio_preprocessor_qwen3a : mtmd_audio_preprocessor {
mtmd_audio_preprocessor_qwen3a(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
void initialize() override;
bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) const override;
private:
mtmd_audio_cache cache;
};
struct mtmd_audio_preprocessor_dots3note : mtmd_audio_preprocessor {
mtmd_audio_preprocessor_dots3note(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
void initialize() override;
bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) const override;
private:
mtmd_audio_cache cache;
};
struct mtmd_audio_preprocessor_mimo_audio : mtmd_audio_preprocessor {
mtmd_audio_preprocessor_mimo_audio(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
void initialize() override;
bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) const override;
private:
mtmd_audio_cache cache;
};
struct mtmd_audio_preprocessor_qwen3tts_spk : mtmd_audio_preprocessor {
mtmd_audio_preprocessor_qwen3tts_spk(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
void initialize() override;
bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) const override;
private:
mtmd_audio_cache cache;
};
// mimi convolves the waveform directly, so this only pads it to a whole number of frames
struct mtmd_audio_preprocessor_pockettts : mtmd_audio_preprocessor {
mtmd_audio_preprocessor_pockettts(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
void initialize() override {}
bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) const override;
};
struct mtmd_audio_preprocessor_parakeet : mtmd_audio_preprocessor {
mtmd_audio_preprocessor_parakeet(clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) { }
void initialize() override;
bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) const override;
private:
mtmd_audio_cache cache;
static void worker_thread(int ith, const float * window_func, int window_size,
const std::vector<float> & samples, int n_samples,
int frame_size, int frame_step, int n_threads,
int n_fft_bins,
const mtmd_audio_cache & cache, mtmd_audio_mel & mel);
};
//
// streaming ISTFT - converts spectrogram frames back to audio one frame at a time
//
struct mtmd_audio_streaming_istft {
mtmd_audio_streaming_istft(int n_fft, int hop_length);
// reset streaming state
void reset();
// process a single STFT frame (streaming)
// frame_spectrum: [n_fft_bins x 2] interleaved real/imag
// returns: up to hop_length samples
std::vector<float> process_frame(const float * frame_spectrum);
// flush remaining samples at end of stream
std::vector<float> flush();
private:
int n_fft;
int hop_length;
int n_fft_bins;
// Own cache for output processing
mtmd_audio_cache cache;
// Streaming state
std::vector<float> overlap_buffer;
std::vector<float> window_sum_buffer;
int padding_to_remove;
// Working buffers for IFFT
std::vector<float> ifft_in;
std::vector<float> ifft_out;
};