mirror of
https://github.com/ggml-org/llama.cpp.git
synced 2026-09-07 05:21:12 +02:00
6e62ba5384
* adapt the api * text model ok * working impl, need verify and clean up * mtmd: build the pocket-tts transposed convolutions as GEMM + col2im ggml_conv_transpose_1d has no grouped mode, so the depthwise upsample was built as one convolution and one concat per channel, which floods the graph with small nodes and makes kernel launches dominate the decoder. Fold both cases into the column form the seanet decoder already needs: the general case reshapes the kernel to [IC, K * OC] and matmuls it with the input, the depthwise case batches a matmul over the channels so a step scales its own kernel. A single col2im_1d then scatter-adds the columns back to the signal, with the same shape as before, so the overlap-add tail, the streaming state and the bias are untouched. Generation time per frame drops by 80% on CUDA and by 50% on CPU. The output matches the previous implementation sample for sample, with a correlation of 0.999994 and identical frame counts. * flow_temp + frames_after_eos * chunking * mtmd: carry the remaining pocket-tts per-pack settings The language packs also tune the end-of-speech padding and the padding of short prompts, next to the temperature already carried in the mmproj: french_24l asks for 8 tail frames instead of the guessed 3, english_2026-01 asks for short prompts to be padded with spaces. Write both in the mmproj as clip.gen.audio.frames_after_eos and clip.gen.audio.pad_short_text, keyed on the pack in the conversion script like the temperature. The loader keeps them optional, so a mmproj without them behaves as before. Map semicolons to commas for every pack instead, the reference only asks for it on three of them and it costs nothing elsewhere. Existing mmproj files must be converted again to carry the two keys. On a long french text the port now lands within 2% of the reference: 22.96s against 23.44s, with the same peak level and the same amount of silence. * clip.gen.audio.model_variant * clean up code comments * nit: drop the dead flow_temp hparam, the pack table holds the default * update docs * address security problems * less invasive base.py * lint * add mtmd_gen_inp_default * add docs * rm gen_flow_temp --------- Co-authored-by: Pascal <admin@serveurperso.com>
188 lines
6.4 KiB
C++
188 lines
6.4 KiB
C++
#pragma once
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#include "ggml.h"
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#include "clip-model.h"
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#include <cstdint>
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#include <vector>
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#include <string>
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#define MTMD_INTERNAL_HEADER
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struct mtmd_audio_mel {
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int64_t n_len;
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int64_t n_len_org;
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int64_t n_mel;
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std::vector<float> data;
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};
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struct mtmd_audio_mel_filters {
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int64_t n_mel;
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int64_t n_fft;
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std::vector<float> data;
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};
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// cache for audio processing, each processor instance owns its own cache
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struct mtmd_audio_cache {
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std::vector<float> sin_vals;
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std::vector<float> cos_vals;
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std::vector<float> hann_window;
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mtmd_audio_mel_filters filters;
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void fill_sin_cos_table(uint32_t n);
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void fill_hann_window(uint32_t length, bool periodic);
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// Build mel filterbank matrix [n_mel × n_fft_bins] at runtime.
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// n_fft_bins must be (N_fft / 2 + 1). Example: if N_fft=512 -> n_fft_bins=257.
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void fill_mel_filterbank_matrix(int64_t n_mel,
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int64_t n_fft,
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int sample_rate, // e.g. 16000
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float fmin = 0.0f, // e.g. 0.0
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float fmax = -1.0f, // e.g. sr/2; pass -1 for auto
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bool slaney_area_norm = true,
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float scale = 1.0f,
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bool use_htk = false
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);
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};
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struct mtmd_audio_preprocessor {
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const clip_hparams & hparams;
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mtmd_audio_preprocessor(const clip_ctx * ctx): hparams(*clip_get_hparams(ctx)) {}
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virtual ~mtmd_audio_preprocessor() = default;
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virtual void initialize() = 0; // NOT thread-safe
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virtual bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) = 0;
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};
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struct mtmd_audio_preprocessor_whisper : mtmd_audio_preprocessor {
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mtmd_audio_preprocessor_whisper(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
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void initialize() override;
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bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) override;
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private:
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mtmd_audio_cache cache;
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};
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struct mtmd_audio_preprocessor_conformer : mtmd_audio_preprocessor {
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mtmd_audio_preprocessor_conformer(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
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void initialize() override;
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bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) override;
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private:
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mtmd_audio_cache cache;
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};
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struct mtmd_audio_preprocessor_granite_speech : mtmd_audio_preprocessor {
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mtmd_audio_preprocessor_granite_speech(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
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void initialize() override;
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bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) override;
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private:
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mtmd_audio_cache cache;
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};
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struct mtmd_audio_preprocessor_gemma4a : mtmd_audio_preprocessor {
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mtmd_audio_preprocessor_gemma4a(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
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void initialize() override;
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bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) override;
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private:
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mtmd_audio_cache cache;
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};
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struct mtmd_audio_preprocessor_gemma4ua : mtmd_audio_preprocessor {
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mtmd_audio_preprocessor_gemma4ua(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
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void initialize() override;
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bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) override;
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};
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struct mtmd_audio_preprocessor_qwen3a : mtmd_audio_preprocessor {
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mtmd_audio_preprocessor_qwen3a(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
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void initialize() override;
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bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) override;
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private:
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mtmd_audio_cache cache;
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};
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struct mtmd_audio_preprocessor_mimo_audio : mtmd_audio_preprocessor {
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mtmd_audio_preprocessor_mimo_audio(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
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void initialize() override;
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bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) override;
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private:
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mtmd_audio_cache cache;
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};
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struct mtmd_audio_preprocessor_qwen3tts_spk : mtmd_audio_preprocessor {
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mtmd_audio_preprocessor_qwen3tts_spk(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
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void initialize() override;
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bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) override;
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private:
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mtmd_audio_cache cache;
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};
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// mimi convolves the waveform directly, so this only pads it to a whole number of frames
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struct mtmd_audio_preprocessor_pockettts : mtmd_audio_preprocessor {
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mtmd_audio_preprocessor_pockettts(const clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) {}
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void initialize() override {}
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bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) override;
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};
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struct mtmd_audio_preprocessor_parakeet : mtmd_audio_preprocessor {
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mtmd_audio_preprocessor_parakeet(clip_ctx * ctx) : mtmd_audio_preprocessor(ctx) { }
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void initialize() override;
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bool preprocess(const float * samples, size_t n_samples, std::vector<mtmd_audio_mel> & output) override;
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private:
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mtmd_audio_cache cache;
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static void worker_thread(int ith, const float * window_func, int window_size,
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const std::vector<float> & samples, int n_samples,
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int frame_size, int frame_step, int n_threads,
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int n_fft_bins,
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const mtmd_audio_cache & cache, mtmd_audio_mel & mel);
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};
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//
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// streaming ISTFT - converts spectrogram frames back to audio one frame at a time
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//
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struct mtmd_audio_streaming_istft {
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mtmd_audio_streaming_istft(int n_fft, int hop_length);
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// reset streaming state
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void reset();
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// process a single STFT frame (streaming)
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// frame_spectrum: [n_fft_bins x 2] interleaved real/imag
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// returns: up to hop_length samples
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std::vector<float> process_frame(const float * frame_spectrum);
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// flush remaining samples at end of stream
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std::vector<float> flush();
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private:
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int n_fft;
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int hop_length;
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int n_fft_bins;
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// Own cache for output processing
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mtmd_audio_cache cache;
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// Streaming state
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std::vector<float> overlap_buffer;
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std::vector<float> window_sum_buffer;
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int padding_to_remove;
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// Working buffers for IFFT
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std::vector<float> ifft_in;
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std::vector<float> ifft_out;
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};
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