Commit Graph

15 Commits

Author SHA1 Message Date
Daniel Bevenius
7e1e28cae3 mtmd : add Nemotron 3 Nano Omni support (parakeet) (#22520)
* mtmd : add Nemotron 3 Nano Omni support (parakeet)

This commit adds support for the subsampling and encoder part of
Nemotron Nemo 3 omni model.

The Parakeet subsampling/encoder were taken from parakeet.cpp which
is currently a pull request against whisper.cpp. I've tried to copy the
code a close as possible to hopefully enable easy patching between the
these two project later.

Refs: https://github.com/ggml-org/whisper.cpp/pull/3735

* mtmd : generate rel pos tensor in graph instead of in conversion [no ci]

This commit removes the generation of the relative positional tensor in
the model conversion script and instead computes it in the encoder
graph. This is only done for the window of positions required for the
current audio sample.

* mtmd : add clip_get_model to clip API [no ci]

This commit adds a function to get access to the clip_model. It also
removes the two functions clip_get_mel_filter_tensor, and
clip_get_window_tensor(const struct clip_ctx * ctx) which can now use
clip_get_model to access the model tensors that it needs.

* mtmd : read mel_filters and window into hparams

* mtmd : use set_input_f32 lambda [no ci]

* mtmd : add better asserts for mel_filters and hann window [no ci]

* mtmd : add missing size_t cast

* mtmd : change type of pad to size_t

* mtmd : zero initialize samples_padded

* mtmd : remove unsued ctx member from parakeet preprocessor

* mtmd : make log_mel_spectrogram_parakeet_worker_thread private static

* mtmd : sync/update parakeeet impl with latest whisper.cpp

This commit updates the parakeet code in mtmd to reflect the latest
updates to parakeet.cpp in whisper.cpp.

A follow up commit will address the currently hardcoded dw_pad and see
if we can add n_conv_kernel as a model metadata field.

* mtmd : add audio_conv_kernel_size to model conversion

This commit updates the model conversion to read the conv_kernel_size
field from the sound_config section of the models config.json file.
It then uses this field instead of the hardcoded values in parakeet.cpp.

* mtmd : cleanup [no ci]

* conversion : call super().filter_tensors [no ci]

* do not discard result of super filter_tensors

* mtmd : use build_mm instead of ggml_mul_mat

* mtmd : use build_ffn

* mtmd : move and reuse get_vector lambda

* mtmd : use build_inp_raw for parakeet

* mtmd : throw exception in get_scalar instead of assert

* mtmd : fix std::min call

* mtmt : use .c_str in throw clause in get_vector

* mtmd : check for F32 type and non-empty tensor in get_vector

The get_vector lambda is used by get_scalar but also standalone to read
in the mel_filters and the window data. Therefor we are not checking
for 1D tensors but allowing multiple dimensions. We do have a check in
get_scalar to verify the size of the vector.

* mtmd : replace hardcoded 1101 for n_tokens_real

* mtmd : assert subsampling_factor is 8

This commit adds an assert of the parakeet subsampling factor to check
that it is 8.

The motivation for this is that this model currently has three
convolutions with a stride of 2. If the underlying model updates the
subsampling factor these convolution operations will need to be updated
and this will produce and error if this occurs.

* mtmd : remove unused ggml_tensors attn_pos_w and mm_norm_w

* mtmd : remove single thread path

This commit removes the single thread path which was a left over from
the original parakeet.cpp where n_threads is configurable.

* fix some security issues

---------

Co-authored-by: Sigbjørn Skjæret <1629204+CISC@users.noreply.github.com>
Co-authored-by: Xuan Son Nguyen <son@huggingface.co>
2026-07-28 17:20:25 +02:00
Xuan-Son Nguyen
1cbfd19883 mtmd: support MiMo-V2.5 audio input (RVQ-based model) (#26190)
* gguf converter for mimo audio

* fix conv

* cpp impl

* nits

* nits 2
2026-07-27 23:17:09 +02:00
Xuan-Son Nguyen
e2e7a9b2d0 mtmd: several bug fixes (#24784)
* mtmd: several bug fixes

* fix build

* fix gemma4ua

* add sanity check in get_u32()

* fix build (2)

* area() avoid overflow
2026-06-19 12:18:36 +02:00
Xuan-Son Nguyen
a731805ced mtmd, model: allow skip build_vit() (#24077)
* add model

* nits
2026-06-03 17:10:35 +02:00
Xuan-Son Nguyen
72e60f500d mtmd: add chunks and fix preproc for qwen3a (#23073)
* mtmd: add chunks and fix preproc for qwen3a

* add attn_mask

* limit mtmd_chunk size (avoid blow up memory)

* correct audio tokens

* re-order the set_input case

* remove attn_mask
2026-05-15 19:32:47 +02:00
Pascal
cc97e45a14 mtmd: fix whisper audio tail truncation by exposing padded buffer to FFT (#22770) 2026-05-07 14:01:01 +02:00
Yakine Tahtah
a00e47e422 mtmd: add granite-speech support (ibm-granite/granite-4.0-1b-speech) (#22101)
* mtmd: add granite-speech support (ibm-granite/granite-4.0-1b-speech)

Conformer encoder with Shaw relative position encoding,
QFormer projector, log-mel spectrogram with frame stacking.

Encoder uses GLU gating, folded batch norm, and SSM depthwise
conv. QFormer compresses encoder output via windowed
cross-attention (window=15, queries=3) into the LLM embedding
space.

Audio preprocessing: reflect-padded STFT, 80-bin mel filterbank,
dynamic range compression, 2x frame stacking (80->160 mel).

GGUF converter handles batch norm folding at export time,
fused K/V split, and Conv1d weight reshaping.

Tested against HF transformers reference: token-for-token match
on 30s/60s audio clips with greedy decoding.

* mtmd: rename gs_ prefixed tensors to generic/architecture names

* mtmd: use tensor_mapping.py for all granite_speech tensors

* convert: fold GraniteSpeechTextModel into GraniteModel

* mtmd: replace n_layer hack with explicit has_standard_layers flag

* mtmd: replace hardcoded magic numbers with GGUF hparams for granite speech

* mtmd: align KEY_A_ define spacing

* convert: register GraniteModel for GraniteSpeechForConditionalGeneration

* convert: fix ty type-check for GraniteSpeechMmprojModel registration

* mtmd: align TN_ define spacing

* mtmd: use generic layer loop for granite speech tensor loading

* mtmd: merge qformer_proj_layer into clip_layer

* mtmd: granite_speech remove redundant ggml_build_forward_expand on inputs

* mtmd: granite_speech add comment explaining why build_attn is not used

* mtmd: granite_speech hard-code eps in cpp, remove from GGUF metadata

* gguf: add spacing between granite_speech tensor mapping blocks

* mtmd: make generic audio layer_norm_eps read optional

* mtmd: granite_speech keep encoder eps in GGUF, only hard-code projector eps

* mtmd: align defines and struct fields in clip-impl.h and clip-model.h

* mtmd: fix alignment and ordering issues across granite speech files

* convert: granite_speech use filter_tensors instead of modify_tensors for skipping
2026-05-06 14:40:59 +02:00
Stephen Cox
547765a93e mtmd: add Gemma 4 audio conformer encoder support (#21421)
* mtmd: add Gemma 4 audio conformer encoder support

Add audio processing for Gemma 4 E2B/E4B via a USM-style Conformer.

Architecture:
- 12-layer Conformer: FFN → Self-Attention → Causal Conv1D → FFN → Norm
- Subsampling Conv Projection: 2x Conv2D(stride=2) with LayerNorm
- Full self-attention with sinusoidal RPE and sliding window mask (24)
- Logit softcapping at 50.0, ClippableLinear clamping
- Output: 1024 → 1536 → RMSNorm → multimodal embedder

Mel preprocessing (dedicated mtmd_audio_preprocessor_gemma4a):
- HTK mel scale, 128 bins, magnitude STFT, mel_floor=1e-3
- Standard periodic Hann window (320 samples), zero-padded to FFT size
- Semicausal left-padding (frame_length/2 samples)
- Frame count matched to PyTorch (unfold formula)
- No pre-emphasis, no Whisper-style normalization
- Mel cosine similarity vs PyTorch: 0.9998

Key fixes:
- Tensor loading dedup: prevent get_tensor() from creating duplicate
  entries in ctx_data. Fixed with std::set guard.
- ClippableLinear clamp_info loading moved after per-layer tensors.
- Sliding window mask (24 positions) matching PyTorch context_size.
- Skip Whisper normalization for Gemma4 mel output.

Tested on E2B and E4B with CPU and Vulkan backends.
Transcribes: "Glad to see things are going well and business is starting
to pick up" (matching ground truth).

Ref: #21325
2026-04-12 14:15:26 +02:00
Xuan-Son Nguyen
871f1a2d2f mtmd: add more sanity checks (#21047) 2026-03-27 11:00:52 +01:00
Marcel Petrick
92f7da00b4 chore : correct typos [no ci] (#20041)
* fix(docs): correct typos found during code review

Non-functional changes only:
- Fixed minor spelling mistakes in comments
- Corrected typos in user-facing strings
- No variables, logic, or functional code was modified.

Signed-off-by: Marcel Petrick <mail@marcelpetrick.it>

* Update docs/backend/CANN.md

Co-authored-by: Aaron Teo <taronaeo@gmail.com>

* Revert "Auxiliary commit to revert individual files from 846d1c301281178efbc6ce6060ad34c1ebe45af8"

This reverts commit 02fcf0c7db661d5ff3eff96b2b2db9fdb7213256.

* Update tests/test-backend-ops.cpp

Co-authored-by: Sigbjørn Skjæret <sigbjorn.skjaeret@scala.com>

* Update tests/test-backend-ops.cpp

Co-authored-by: Sigbjørn Skjæret <sigbjorn.skjaeret@scala.com>

---------

Signed-off-by: Marcel Petrick <mail@marcelpetrick.it>
Co-authored-by: Aaron Teo <taronaeo@gmail.com>
Co-authored-by: Sigbjørn Skjæret <sigbjorn.skjaeret@scala.com>
2026-03-05 08:50:21 +01:00
Tarek Dakhran
ccbc84a537 mtmd: mtmd_audio_streaming_istft (#18645)
Change is decoupled from https://github.com/ggml-org/llama.cpp/pull/18641.

[LFM2.5-Audio-1.5B](https://huggingface.co/LiquidAI/LFM2.5-Audio-1.5B)
needs streaming istft for generating output audio.

* add streaming ISTFT class (`mtmd_audio_streaming_istft`) with overlap-add for audio reconstruction
* replace global audio cache with per-instance cache, the model requires
  two independent caches, for preprocessing (audio input) and for istft
  (audio output).
* unified templated FFT/IFFT implementation supporting both forward and inverse transforms
2026-01-06 21:00:29 +01:00
Xuan-Son Nguyen
8ea958d4d9 model : add ASR support for LFM2-Audio-1.5B (conformer) (#18106)
* ASR with LFM2-Audio-1.5B

* Set rope_theta

* Fix comment

* Remove rope_theta setting

* Address PR feedback

* rename functions to conformer

* remove some redundant ggml_cont

* fix missing tensor

* add prefix "a." for conv tensors

* remove redundant reshape

* clean up

* add test model

---------

Co-authored-by: Tarek Dakhran <tarek@liquid.ai>
2025-12-19 00:18:01 +01:00
Xuan-Son Nguyen
96a181a933 mtmd: refactor audio preprocessing (#17978)
* mtmd: refactor audio preprocessing

* refactor

Co-authored-by: Tarek <tdakhran@users.noreply.github.com>

* wip

* wip (2)

* improve constructor

* fix use_natural_log

* fix padding for short input

* clean up

* remove need_chunking

---------

Co-authored-by: Tarek <tdakhran@users.noreply.github.com>
2025-12-15 14:16:52 +01:00
Xuan-Son Nguyen
10961339b2 mtmd : move helpers to dedicated library (⚠️ breaking change) (#13866)
* mtmd : move helpers to dedicated library

* fix server build

* rm leftover cmakelist code
2025-05-28 22:35:22 +02:00
Xuan-Son Nguyen
797990c4bc mtmd : add ultravox audio input (#13623)
* convert ok, load ok

* warmup ok

* test

* still does not work?

* fix padding

* temporary give up

* fix merge conflict

* build_ultravox()

* rm test

* fix merge conflict

* add necessary mtmd APIs

* first working version (only 4s of audio)

* will this monster compile?

* fix compile

* please compile

* fPIC

* fix windows

* various fixes

* clean up audio_helpers

* fix conversion

* add some debug stuff

* long audio input ok

* adapt the api

* add --audio arg

* final touch UX

* add miniaudio to readme

* fix typo

* refactor kv metadata

* mtmd_default_marker()
2025-05-22 20:42:48 +02:00