Upload folder using huggingface_hub

train-higgs-audio/.github/workflows/test.yml

@@ -0,0 +1,20 @@
+name: Unit Test
+on:
+  push:
+    branches: [ main ]
+  pull_request:
+    branches: [ main ]
+
+jobs:
+  lint:
+    name: Lint
+    runs-on: ubuntu-22.04
+    steps:
+      - name: Checkout code 
+        uses: actions/checkout@v4
+
+      - name: Check Code Formatting with Ruff
+        run: |
+            echo "python version: $(python --version)"
+            pip install ruff==0.12.2  # Ensure ruff is installed
+            ruff format --check .

train-higgs-audio/boson_multimodal.egg-info/PKG-INFO

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+Metadata-Version: 2.4
+Name: boson_multimodal
+Version: 0.1.0
+Summary: Higgs Audio
+Home-page: https://github.com/boson-ai/higgs-audio
+Author: Boson AI
+Description-Content-Type: text/markdown
+
+<h1 align="center">Higgs Audio V2: Redefining Expressiveness in Audio Generation</h1>
+
+<div align="center" style="display: flex; justify-content: center; margin-top: 10px;">
+  <a href="https://boson.ai/blog/higgs-audio-v2"><img src='https://img.shields.io/badge/🚀-Launch Blogpost-228B22' style="margin-right: 5px;"></a>
+  <a href="https://boson.ai/demo/tts"><img src="https://img.shields.io/badge/🕹️-Boson%20AI%20Playground-9C276A" style="margin-right: 5px;"></a>
+  <a href="https://huggingface.co/spaces/smola/higgs_audio_v2"><img src="https://img.shields.io/badge/🎮-HF%20Space%20Playground-8A2BE2" style="margin-right: 5px;"></a>
+  <a href="https://huggingface.co/bosonai/higgs-audio-v2-generation-3B-base"><img src="https://img.shields.io/badge/🤗-Checkpoints (3.6B LLM + 2.2B audio adapter)-ED5A22.svg" style="margin-right: 5px;"></a>
+</div>
+
+
+We are open-sourcing Higgs Audio v2, a powerful audio foundation model pretrained on over 10 million hours of audio data and a diverse set of text data. Despite having no post-training or fine-tuning, Higgs Audio v2 excels in expressive audio generation, thanks to its deep language and acoustic understanding.
+
+On [EmergentTTS-Eval](https://github.com/boson-ai/emergenttts-eval-public), it achieves win rates of **75.7%** and **55.7%** over "gpt-4o-mini-tts" on the "Emotions" and "Questions" categories, respectively. It also obtains state-of-the-art performance on traditional TTS benchmarks like Seed-TTS Eval and Emotional Speech Dataset (ESD). Moreover, the model demonstrates capabilities rarely seen in previous systems, including generating natural multi-speaker dialogues in multiple languages, automatic prosody adaptation during narration, melodic humming with the cloned voice, and simultaneous generation of speech and background music.
+
+<p align="center">
+    <img src="figures/emergent-tts-emotions-win-rate.png" width=900>
+</p>
+
+Here's the demo video that shows some of its emergent capabilities (remember to unmute):
+
+<video src="https://github.com/user-attachments/assets/0fd73fad-097f-48a9-9f3f-bc2a63b3818d" type="video/mp4" width="80%" controls>
+</video>
+
+Here's another demo video that show-cases the model's multilingual capability and how it enabled live translation (remember to unmute):
+
+<video src="https://github.com/user-attachments/assets/2b9b01ff-67fc-4bd9-9714-7c7df09e38d6" type="video/mp4" width="80%" controls>
+</video>
+
+## Installation
+
+We recommend to use NVIDIA Deep Learning Container to manage the CUDA environment. Following are two docker images that we have verified:
+- nvcr.io/nvidia/pytorch:25.02-py3
+- nvcr.io/nvidia/pytorch:25.01-py3
+
+Here's an example command for launching a docker container environment. Please also check the [official NVIDIA documentations](https://catalog.ngc.nvidia.com/orgs/nvidia/containers/pytorch).
+
+```bash
+docker run --gpus all --ipc=host --net=host --ulimit memlock=-1 --ulimit stack=67108864 -it --rm nvcr.io/nvidia/pytorch:25.02-py3 bash
+```
+
+### Option 1: Direct installation
+
+
+```bash
+git clone https://github.com/boson-ai/higgs-audio.git
+cd higgs-audio
+
+pip install -r requirements.txt
+pip install -e .
+```
+
+### Option 2: Using venv
+
+```bash
+git clone https://github.com/boson-ai/higgs-audio.git
+cd higgs-audio
+
+python3 -m venv higgs_audio_env
+source higgs_audio_env/bin/activate
+pip install -r requirements.txt
+pip install -e .
+```
+
+
+### Option 3: Using conda
+```bash
+git clone https://github.com/boson-ai/higgs-audio.git
+cd higgs-audio
+
+conda create -n higgs_audio_env python=3.10
+conda activate higgs_audio_env
+pip install -r requirements.txt
+pip install -e .
+```
+
+### Option 4: Using uv
+```bash
+git clone https://github.com/boson-ai/higgs-audio.git
+cd higgs-audio
+
+uv venv --python 3.10
+source .venv/bin/activate
+uv pip install -r requirements.txt
+uv pip install -e .
+```
+
+### Option 5: Using vllm
+
+For advanced usage with higher throughput, we also built OpenAI compatible API server backed by vLLM engine for you to use.
+Please refer to [examples/vllm](./examples/vllm) for more details.
+
+
+## Usage
+
+> [!TIP]
+> For optimal performance, run the generation examples on a machine equipped with GPU with at least 24GB memory!
+
+### Get Started
+
+Here's a basic python snippet to help you get started.
+
+```python
+from boson_multimodal.serve.serve_engine import HiggsAudioServeEngine, HiggsAudioResponse
+from boson_multimodal.data_types import ChatMLSample, Message, AudioContent
+
+import torch
+import torchaudio
+import time
+import click
+
+MODEL_PATH = "bosonai/higgs-audio-v2-generation-3B-base"
+AUDIO_TOKENIZER_PATH = "bosonai/higgs-audio-v2-tokenizer"
+
+system_prompt = (
+    "Generate audio following instruction.\n\n<|scene_desc_start|>\nAudio is recorded from a quiet room.\n<|scene_desc_end|>"
+)
+
+messages = [
+    Message(
+        role="system",
+        content=system_prompt,
+    ),
+    Message(
+        role="user",
+        content="The sun rises in the east and sets in the west. This simple fact has been observed by humans for thousands of years.",
+    ),
+]
+device = "cuda" if torch.cuda.is_available() else "cpu"
+
+serve_engine = HiggsAudioServeEngine(MODEL_PATH, AUDIO_TOKENIZER_PATH, device=device)
+
+output: HiggsAudioResponse = serve_engine.generate(
+    chat_ml_sample=ChatMLSample(messages=messages),
+    max_new_tokens=1024,
+    temperature=0.3,
+    top_p=0.95,
+    top_k=50,
+    stop_strings=["<|end_of_text|>", "<|eot_id|>"],
+)
+torchaudio.save(f"output.wav", torch.from_numpy(output.audio)[None, :], output.sampling_rate)
+```
+
+We also provide a list of examples under [examples](./examples). In the following we highlight a few examples to help you use Higgs Audio v2.
+
+### Zero-Shot Voice Cloning
+Generate audio that sounds similar as the provided [reference audio](./examples/voice_prompts/belinda.wav).
+
+```bash
+python3 examples/generation.py \
+--transcript "The sun rises in the east and sets in the west. This simple fact has been observed by humans for thousands of years." \
+--ref_audio belinda \
+--temperature 0.3 \
+--out_path generation.wav
+```
+
+The generation script will automatically use `cuda:0` if it founds cuda is available. To change the device id, specify `--device_id`:
+
+```bash
+python3 examples/generation.py \
+--transcript "The sun rises in the east and sets in the west. This simple fact has been observed by humans for thousands of years." \
+--ref_audio belinda \
+--temperature 0.3 \
+--device_id 0 \
+--out_path generation.wav
+```
+
+You can also try other voices. Check more example voices in [examples/voice_prompts](./examples/voice_prompts). You can also add your own voice to the folder.
+
+```bash
+python3 examples/generation.py \
+--transcript "The sun rises in the east and sets in the west. This simple fact has been observed by humans for thousands of years." \
+--ref_audio broom_salesman \
+--temperature 0.3 \
+--out_path generation.wav
+```
+
+### Single-speaker Generation with Smart Voice
+If you do not specify reference voice, the model will decide the voice based on the transcript it sees.
+
+```bash
+python3 examples/generation.py \
+--transcript "The sun rises in the east and sets in the west. This simple fact has been observed by humans for thousands of years." \
+--temperature 0.3 \
+--out_path generation.wav
+```
+
+
+### Multi-speaker Dialog with Smart Voice
+Generate multi-speaker dialog. The model will decide the voices based on the transcript it sees.
+
+```bash
+python3 examples/generation.py \
+--transcript examples/transcript/multi_speaker/en_argument.txt \
+--seed 12345 \
+--out_path generation.wav
+```
+
+### Multi-speaker Dialog with Voice Clone
+
+Generate multi-speaker dialog with the voices you picked.
+
+```bash
+python3 examples/generation.py \
+--transcript examples/transcript/multi_speaker/en_argument.txt \
+--ref_audio belinda,broom_salesman \
+--ref_audio_in_system_message \
+--chunk_method speaker \
+--seed 12345 \
+--out_path generation.wav
+```
+
+
+## Technical Details
+<img src="figures/higgs_audio_v2_architecture_combined.png" width=900>
+
+
+Higgs Audio v2 adopts the "generation variant" depicted in the architecture figure above. Its strong performance is driven by three key technical innovations:
+- We developed an automated annotation pipeline that leverages multiple ASR models, sound event classification models, and our in-house audio understanding model. Using this pipeline, we cleaned and annotated 10 million hours audio data, which we refer to as **AudioVerse**. The in-house understanding model is finetuned on top of [Higgs Audio v1 Understanding](https://www.boson.ai/blog/higgs-audio), which adopts the "understanding variant" shown in the architecture figure.
+- We trained a unified audio tokenizer from scratch that captures both semantic and acoustic features. Learn more in the [tokenizer blog](./tech_blogs/TOKENIZER_BLOG.md).
+- We proposed the DualFFN architecture, which enhances the LLM’s ability to model acoustics tokens with minimal computational overhead. See the [architecture blog](./tech_blogs/ARCHITECTURE_BLOG.md).
+
+## Evaluation
+
+Here's the performance of Higgs Audio v2 on four benchmarks,  [Seed-TTS Eval](https://github.com/BytedanceSpeech/seed-tts-eval), [Emotional Speech Dataset (ESD)](https://paperswithcode.com/dataset/esd), [EmergentTTS-Eval](https://arxiv.org/abs/2505.23009), and Multi-speaker Eval:
+
+#### Seed-TTS Eval & ESD
+
+We prompt Higgs Audio v2 with the reference text, reference audio, and target text for zero-shot TTS. We use the standard evaluation metrics from Seed-TTS Eval and ESD.
+
+|                              | SeedTTS-Eval| | ESD   |                 |
+|------------------------------|--------|--------|---------|-------------------|
+|                              | WER ↓ | SIM ↑ | WER ↓ | SIM (emo2vec) ↑ |
+| Cosyvoice2                   | 2.28   | 65.49  | 2.71    | 80.48             |
+| Qwen2.5-omni†                | 2.33   | 64.10  | -       | -                 |
+| ElevenLabs Multilingual V2   | **1.43**   | 50.00  | 1.66    | 65.87             |
+| Higgs Audio v1                | 2.18   | 66.27  | **1.49**    | 82.84             |
+| Higgs Audio v2 (base)         | 2.44   | **67.70**  | 1.78    | **86.13**         |
+
+
+#### EmergentTTS-Eval ("Emotions" and "Questions")
+
+Following the [EmergentTTS-Eval Paper](https://arxiv.org/abs/2505.23009), we report the win-rate over "gpt-4o-mini-tts" with the "alloy" voice. The judge model is Gemini 2.5 Pro.
+
+| Model                              | Emotions (%) ↑ | Questions (%) ↑ |
+|------------------------------------|--------------|----------------|
+| Higgs Audio v2 (base)               | **75.71%**   | **55.71%**         |
+| [gpt-4o-audio-preview†](https://platform.openai.com/docs/models/gpt-4o-audio-preview)       | 61.64%       | 47.85%         |
+| [Hume.AI](https://www.hume.ai/research)                            | 61.60%       | 43.21%         |
+| **BASELINE:** [gpt-4o-mini-tts](https://platform.openai.com/docs/models/gpt-4o-mini-tts)  | 50.00%       | 50.00%         |
+| [Qwen 2.5 Omni†](https://github.com/QwenLM/Qwen2.5-Omni)      | 41.60%       | 51.78%         |
+| [minimax/speech-02-hd](https://replicate.com/minimax/speech-02-hd)               | 40.86%        | 47.32%         |
+| [ElevenLabs Multilingual v2](https://elevenlabs.io/blog/eleven-multilingual-v2)         | 30.35%       | 39.46%         |
+| [DeepGram Aura-2](https://deepgram.com/learn/introducing-aura-2-enterprise-text-to-speech)                    | 29.28%       | 48.21%         |
+| [Sesame csm-1B](https://github.com/SesameAILabs/csm)                      | 15.96%       | 31.78%         |
+
+<sup><sub>'†' means using the strong-prompting method described in the paper.</sub></sup>
+
+
+#### Multi-speaker Eval
+
+We also designed a multi-speaker evaluation benchmark to evaluate the capability of Higgs Audio v2 for multi-speaker dialog generation. The benchmark contains three subsets
+
+- `two-speaker-conversation`: 1000 synthetic dialogues involving two speakers. We fix two reference audio clips to evaluate the model's ability in double voice cloning for utterances ranging from 4 to 10 dialogues between two randomly chosen persona.
+- `small talk (no ref)`: 250 synthetic dialogues curated in the same way as above, but are characterized by short utterances and a limited number of turns (4–6), we do not fix reference audios in this case and this set is designed to evaluate the model's ability to automatically assign appropriate voices to speakers.
+- `small talk (ref)`: 250 synthetic dialogues similar to above, but contains even shorter utterances as this set is meant to include reference clips in it's context, similar to `two-speaker-conversation`.
+
+
+We report the word-error-rate (WER) and the geometric mean between intra-speaker similarity and inter-speaker dis-similarity on these three subsets. Other than Higgs Audio v2, we also evaluated [MoonCast](https://github.com/jzq2000/MoonCast) and [nari-labs/Dia-1.6B-0626](https://huggingface.co/nari-labs/Dia-1.6B-0626), two of the most popular open-source models capable of multi-speaker dialog generation. Results are summarized in the following table. We are not able to run [nari-labs/Dia-1.6B-0626](https://huggingface.co/nari-labs/Dia-1.6B-0626) on our "two-speaker-conversation" subset due to its strict limitation on the length of the utterances and output audio.
+
+|                                                | two-speaker-conversation |                |small talk |                | small talk (no ref) |                |
+| ---------------------------------------------- | -------------- | ------------------ | ---------- | -------------- | ------------------- | -------------- |
+|                                                | WER ↓                      | Mean Sim & Dis-sim ↑ | WER ↓       |  Mean Sim & Dis-sim ↑ | WER ↓               | Mean Sim & Dis-sim ↑ |
+| [MoonCast](https://github.com/jzq2000/MoonCast) | 38.77                    | 46.02         | **8.33**       | 63.68          | 24.65               | 53.94 |
+| [nari-labs/Dia-1.6B-0626](https://huggingface.co/nari-labs/Dia-1.6B-0626)         | \-                       | \-             | 17.62      | 63.15          | 19.46               | **61.14**          |
+| Higgs Audio v2 (base)     | **18.88**                    | **51.95**          | 11.89      | **67.92**              | **14.65**               | 55.28              |
+
+
+## Third-Party Licenses
+
+The `boson_multimodal/audio_processing/` directory contains code derived from third-party repositories, primarily from [xcodec](https://github.com/zhenye234/xcodec). Please see the [`LICENSE`](boson_multimodal/audio_processing/LICENSE) in that directory for complete attribution and licensing information.

train-higgs-audio/boson_multimodal.egg-info/SOURCES.txt

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+README.md
+pyproject.toml
+setup.cfg
+setup.py
+boson_multimodal/__init__.py
+boson_multimodal/constants.py
+boson_multimodal/data_types.py
+boson_multimodal.egg-info/PKG-INFO
+boson_multimodal.egg-info/SOURCES.txt
+boson_multimodal.egg-info/dependency_links.txt
+boson_multimodal.egg-info/top_level.txt
+boson_multimodal/data_collator/__init__.py
+boson_multimodal/data_collator/higgs_audio_collator.py
+boson_multimodal/dataset/__init__.py
+boson_multimodal/dataset/chatml_dataset copy.py
+boson_multimodal/dataset/chatml_dataset.py
+boson_multimodal/model/__init__.py
+boson_multimodal/model/higgs_audio/__init__.py
+boson_multimodal/model/higgs_audio/audio_head.py
+boson_multimodal/model/higgs_audio/common.py
+boson_multimodal/model/higgs_audio/configuration_higgs_audio.py
+boson_multimodal/model/higgs_audio/cuda_graph_runner.py
+boson_multimodal/model/higgs_audio/custom_modules.py
+boson_multimodal/model/higgs_audio/modeling_higgs_audio.py
+boson_multimodal/model/higgs_audio/utils.py

train-higgs-audio/boson_multimodal.egg-info/dependency_links.txt

@@ -0,0 +1 @@
+

train-higgs-audio/boson_multimodal.egg-info/top_level.txt

@@ -0,0 +1 @@
+boson_multimodal

train-higgs-audio/boson_multimodal/__init__.py

@@ -0,0 +1 @@
+from .model.higgs_audio import HiggsAudioConfig, HiggsAudioModel

train-higgs-audio/boson_multimodal/audio_processing/LICENSE

@@ -0,0 +1,51 @@
+Third-Party License Attribution for Audio Processing Module
+===========================================================
+
+This directory contains code derived from multiple open-source projects. 
+The following sections detail the licenses and attributions for third-party code.
+
+## XCodec Repository
+The code in this directory is derived from:
+https://github.com/zhenye234/xcodec
+
+## Individual File Attributions
+
+### Quantization Module (quantization/)
+- Several files contain code derived from Meta Platforms, Inc. and the vector-quantize-pytorch repository
+- Individual files contain their own license headers where applicable
+- The vector-quantize-pytorch portions are licensed under the MIT License
+
+## License Terms
+
+### MIT License (for applicable portions)
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+SOFTWARE.
+
+## Attribution Requirements
+When using this code, please ensure proper attribution to:
+1. The original xcodec repository: https://github.com/zhenye234/xcodec
+2. Any other repositories mentioned in individual file headers
+3. This derivative work and its modifications
+
+## Disclaimer
+This directory contains modified versions of the original code. Please refer to
+the original repositories for the canonical implementations and their specific
+license terms.
+
+For any questions about licensing or attribution, please check the individual
+file headers and the original source repositories. 

train-higgs-audio/boson_multimodal/audio_processing/descriptaudiocodec/dac/model/base.py

@@ -0,0 +1,286 @@
+import math
+from dataclasses import dataclass
+from pathlib import Path
+from typing import Union
+
+import numpy as np
+import torch
+import tqdm
+from audiotools import AudioSignal
+from torch import nn
+
+SUPPORTED_VERSIONS = ["1.0.0"]
+
+
+@dataclass
+class DACFile:
+    codes: torch.Tensor
+
+    # Metadata
+    chunk_length: int
+    original_length: int
+    input_db: float
+    channels: int
+    sample_rate: int
+    padding: bool
+    dac_version: str
+
+    def save(self, path):
+        artifacts = {
+            "codes": self.codes.numpy().astype(np.uint16),
+            "metadata": {
+                "input_db": self.input_db.numpy().astype(np.float32),
+                "original_length": self.original_length,
+                "sample_rate": self.sample_rate,
+                "chunk_length": self.chunk_length,
+                "channels": self.channels,
+                "padding": self.padding,
+                "dac_version": SUPPORTED_VERSIONS[-1],
+            },
+        }
+        path = Path(path).with_suffix(".dac")
+        with open(path, "wb") as f:
+            np.save(f, artifacts)
+        return path
+
+    @classmethod
+    def load(cls, path):
+        artifacts = np.load(path, allow_pickle=True)[()]
+        codes = torch.from_numpy(artifacts["codes"].astype(int))
+        if artifacts["metadata"].get("dac_version", None) not in SUPPORTED_VERSIONS:
+            raise RuntimeError(f"Given file {path} can't be loaded with this version of descript-audio-codec.")
+        return cls(codes=codes, **artifacts["metadata"])
+
+
+class CodecMixin:
+    @property
+    def padding(self):
+        if not hasattr(self, "_padding"):
+            self._padding = True
+        return self._padding
+
+    @padding.setter
+    def padding(self, value):
+        assert isinstance(value, bool)
+
+        layers = [l for l in self.modules() if isinstance(l, (nn.Conv1d, nn.ConvTranspose1d))]
+
+        for layer in layers:
+            if value:
+                if hasattr(layer, "original_padding"):
+                    layer.padding = layer.original_padding
+            else:
+                layer.original_padding = layer.padding
+                layer.padding = tuple(0 for _ in range(len(layer.padding)))
+
+        self._padding = value
+
+    def get_delay(self):
+        # Any number works here, delay is invariant to input length
+        l_out = self.get_output_length(0)
+        L = l_out
+
+        layers = []
+        for layer in self.modules():
+            if isinstance(layer, (nn.Conv1d, nn.ConvTranspose1d)):
+                layers.append(layer)
+
+        for layer in reversed(layers):
+            d = layer.dilation[0]
+            k = layer.kernel_size[0]
+            s = layer.stride[0]
+
+            if isinstance(layer, nn.ConvTranspose1d):
+                L = ((L - d * (k - 1) - 1) / s) + 1
+            elif isinstance(layer, nn.Conv1d):
+                L = (L - 1) * s + d * (k - 1) + 1
+
+            L = math.ceil(L)
+
+        l_in = L
+
+        return (l_in - l_out) // 2
+
+    def get_output_length(self, input_length):
+        L = input_length
+        # Calculate output length
+        for layer in self.modules():
+            if isinstance(layer, (nn.Conv1d, nn.ConvTranspose1d)):
+                d = layer.dilation[0]
+                k = layer.kernel_size[0]
+                s = layer.stride[0]
+
+                if isinstance(layer, nn.Conv1d):
+                    L = ((L - d * (k - 1) - 1) / s) + 1
+                elif isinstance(layer, nn.ConvTranspose1d):
+                    L = (L - 1) * s + d * (k - 1) + 1
+
+                L = math.floor(L)
+        return L
+
+    @torch.no_grad()
+    def compress(
+        self,
+        audio_path_or_signal: Union[str, Path, AudioSignal],
+        win_duration: float = 1.0,
+        verbose: bool = False,
+        normalize_db: float = -16,
+        n_quantizers: int = None,
+    ) -> DACFile:
+        """Processes an audio signal from a file or AudioSignal object into
+        discrete codes. This function processes the signal in short windows,
+        using constant GPU memory.
+
+        Parameters
+        ----------
+        audio_path_or_signal : Union[str, Path, AudioSignal]
+            audio signal to reconstruct
+        win_duration : float, optional
+            window duration in seconds, by default 5.0
+        verbose : bool, optional
+            by default False
+        normalize_db : float, optional
+            normalize db, by default -16
+
+        Returns
+        -------
+        DACFile
+            Object containing compressed codes and metadata
+            required for decompression
+        """
+        audio_signal = audio_path_or_signal
+        if isinstance(audio_signal, (str, Path)):
+            audio_signal = AudioSignal.load_from_file_with_ffmpeg(str(audio_signal))
+
+        self.eval()
+        original_padding = self.padding
+        original_device = audio_signal.device
+
+        audio_signal = audio_signal.clone()
+        original_sr = audio_signal.sample_rate
+
+        resample_fn = audio_signal.resample
+        loudness_fn = audio_signal.loudness
+
+        # If audio is > 10 minutes long, use the ffmpeg versions
+        if audio_signal.signal_duration >= 10 * 60 * 60:
+            resample_fn = audio_signal.ffmpeg_resample
+            loudness_fn = audio_signal.ffmpeg_loudness
+
+        original_length = audio_signal.signal_length
+        resample_fn(self.sample_rate)
+        input_db = loudness_fn()
+
+        if normalize_db is not None:
+            audio_signal.normalize(normalize_db)
+        audio_signal.ensure_max_of_audio()
+
+        nb, nac, nt = audio_signal.audio_data.shape
+        audio_signal.audio_data = audio_signal.audio_data.reshape(nb * nac, 1, nt)
+        win_duration = audio_signal.signal_duration if win_duration is None else win_duration
+
+        if audio_signal.signal_duration <= win_duration:
+            # Unchunked compression (used if signal length < win duration)
+            self.padding = True
+            n_samples = nt
+            hop = nt
+        else:
+            # Chunked inference
+            self.padding = False
+            # Zero-pad signal on either side by the delay
+            audio_signal.zero_pad(self.delay, self.delay)
+            n_samples = int(win_duration * self.sample_rate)
+            # Round n_samples to nearest hop length multiple
+            n_samples = int(math.ceil(n_samples / self.hop_length) * self.hop_length)
+            hop = self.get_output_length(n_samples)
+
+        codes = []
+        range_fn = range if not verbose else tqdm.trange
+
+        for i in range_fn(0, nt, hop):
+            x = audio_signal[..., i : i + n_samples]
+            x = x.zero_pad(0, max(0, n_samples - x.shape[-1]))
+
+            audio_data = x.audio_data.to(self.device)
+            audio_data = self.preprocess(audio_data, self.sample_rate)
+            _, c, _, _, _ = self.encode(audio_data, n_quantizers)
+            codes.append(c.to(original_device))
+            chunk_length = c.shape[-1]
+
+        codes = torch.cat(codes, dim=-1)
+
+        dac_file = DACFile(
+            codes=codes,
+            chunk_length=chunk_length,
+            original_length=original_length,
+            input_db=input_db,
+            channels=nac,
+            sample_rate=original_sr,
+            padding=self.padding,
+            dac_version=SUPPORTED_VERSIONS[-1],
+        )
+
+        if n_quantizers is not None:
+            codes = codes[:, :n_quantizers, :]
+
+        self.padding = original_padding
+        return dac_file
+
+    @torch.no_grad()
+    def decompress(
+        self,
+        obj: Union[str, Path, DACFile],
+        verbose: bool = False,
+    ) -> AudioSignal:
+        """Reconstruct audio from a given .dac file
+
+        Parameters
+        ----------
+        obj : Union[str, Path, DACFile]
+            .dac file location or corresponding DACFile object.
+        verbose : bool, optional
+            Prints progress if True, by default False
+
+        Returns
+        -------
+        AudioSignal
+            Object with the reconstructed audio
+        """
+        self.eval()
+        if isinstance(obj, (str, Path)):
+            obj = DACFile.load(obj)
+
+        original_padding = self.padding
+        self.padding = obj.padding
+
+        range_fn = range if not verbose else tqdm.trange
+        codes = obj.codes
+        original_device = codes.device
+        chunk_length = obj.chunk_length
+        recons = []
+
+        for i in range_fn(0, codes.shape[-1], chunk_length):
+            c = codes[..., i : i + chunk_length].to(self.device)
+            z = self.quantizer.from_codes(c)[0]
+            r = self.decode(z)
+            recons.append(r.to(original_device))
+
+        recons = torch.cat(recons, dim=-1)
+        recons = AudioSignal(recons, self.sample_rate)
+
+        resample_fn = recons.resample
+        loudness_fn = recons.loudness
+
+        # If audio is > 10 minutes long, use the ffmpeg versions
+        if recons.signal_duration >= 10 * 60 * 60:
+            resample_fn = recons.ffmpeg_resample
+            loudness_fn = recons.ffmpeg_loudness
+
+        recons.normalize(obj.input_db)
+        resample_fn(obj.sample_rate)
+        recons = recons[..., : obj.original_length]
+        loudness_fn()
+        recons.audio_data = recons.audio_data.reshape(-1, obj.channels, obj.original_length)
+
+        self.padding = original_padding
+        return recons

train-higgs-audio/boson_multimodal/audio_processing/descriptaudiocodec/dac/model/dac.py

@@ -0,0 +1,365 @@
+import math
+from typing import List
+from typing import Union
+
+import numpy as np
+import torch
+from audiotools import AudioSignal
+from audiotools.ml import BaseModel
+from torch import nn
+
+from .base import CodecMixin
+from dac.nn.layers import Snake1d
+from dac.nn.layers import WNConv1d
+from dac.nn.layers import WNConvTranspose1d
+from dac.nn.quantize import ResidualVectorQuantize
+
+
+def init_weights(m):
+    if isinstance(m, nn.Conv1d):
+        nn.init.trunc_normal_(m.weight, std=0.02)
+        nn.init.constant_(m.bias, 0)
+
+
+class ResidualUnit(nn.Module):
+    def __init__(self, dim: int = 16, dilation: int = 1):
+        super().__init__()
+        pad = ((7 - 1) * dilation) // 2
+        self.block = nn.Sequential(
+            Snake1d(dim),
+            WNConv1d(dim, dim, kernel_size=7, dilation=dilation, padding=pad),
+            Snake1d(dim),
+            WNConv1d(dim, dim, kernel_size=1),
+        )
+
+    def forward(self, x):
+        y = self.block(x)
+        pad = (x.shape[-1] - y.shape[-1]) // 2
+        if pad > 0:
+            x = x[..., pad:-pad]
+        return x + y
+
+
+class EncoderBlock(nn.Module):
+    def __init__(self, dim: int = 16, stride: int = 1):
+        super().__init__()
+        self.block = nn.Sequential(
+            ResidualUnit(dim // 2, dilation=1),
+            ResidualUnit(dim // 2, dilation=3),
+            ResidualUnit(dim // 2, dilation=9),
+            Snake1d(dim // 2),
+            WNConv1d(
+                dim // 2,
+                dim,
+                kernel_size=2 * stride,
+                stride=stride,
+                padding=math.ceil(stride / 2),
+            ),
+        )
+
+    def forward(self, x):
+        return self.block(x)
+
+
+class Encoder(nn.Module):
+    def __init__(
+        self,
+        d_model: int = 64,
+        strides: list = [2, 4, 8, 8],
+        d_latent: int = 256,
+    ):
+        super().__init__()
+        # Create first convolution
+        self.block = [WNConv1d(1, d_model, kernel_size=7, padding=3)]
+
+        # Create EncoderBlocks that double channels as they downsample by `stride`
+        for stride in strides:
+            d_model *= 2
+            self.block += [EncoderBlock(d_model, stride=stride)]
+
+        # Create last convolution
+        self.block += [
+            Snake1d(d_model),
+            WNConv1d(d_model, d_latent, kernel_size=3, padding=1),
+        ]
+
+        # Wrap black into nn.Sequential
+        self.block = nn.Sequential(*self.block)
+        self.enc_dim = d_model
+
+    def forward(self, x):
+        return self.block(x)
+
+
+class DecoderBlock(nn.Module):
+    def __init__(self, input_dim: int = 16, output_dim: int = 8, stride: int = 1, out_pad=0):
+        super().__init__()
+        self.block = nn.Sequential(
+            Snake1d(input_dim),
+            WNConvTranspose1d(
+                input_dim,
+                output_dim,
+                kernel_size=2 * stride,
+                stride=stride,
+                padding=math.ceil(stride / 2),
+                output_padding=stride % 2,  # out_pad,
+            ),
+            ResidualUnit(output_dim, dilation=1),
+            ResidualUnit(output_dim, dilation=3),
+            ResidualUnit(output_dim, dilation=9),
+        )
+
+    def forward(self, x):
+        return self.block(x)
+
+
+class Decoder(nn.Module):
+    def __init__(
+        self,
+        input_channel,
+        channels,
+        rates,
+        d_out: int = 1,
+    ):
+        super().__init__()
+
+        # Add first conv layer
+        layers = [WNConv1d(input_channel, channels, kernel_size=7, padding=3)]
+
+        # Add upsampling + MRF blocks
+        for i, stride in enumerate(rates):
+            input_dim = channels // 2**i
+            output_dim = channels // 2 ** (i + 1)
+            if i == 1:
+                out_pad = 1
+            else:
+                out_pad = 0
+            layers += [DecoderBlock(input_dim, output_dim, stride, out_pad)]
+
+        # Add final conv layer
+        layers += [
+            Snake1d(output_dim),
+            WNConv1d(output_dim, d_out, kernel_size=7, padding=3),
+            # nn.Tanh(),
+        ]
+
+        self.model = nn.Sequential(*layers)
+
+    def forward(self, x):
+        return self.model(x)
+
+
+class DAC(BaseModel, CodecMixin):
+    def __init__(
+        self,
+        encoder_dim: int = 64,
+        encoder_rates: List[int] = [2, 4, 8, 8],
+        latent_dim: int = None,
+        decoder_dim: int = 1536,
+        decoder_rates: List[int] = [8, 8, 4, 2],
+        n_codebooks: int = 9,
+        codebook_size: int = 1024,
+        codebook_dim: Union[int, list] = 8,
+        quantizer_dropout: bool = False,
+        sample_rate: int = 44100,
+    ):
+        super().__init__()
+
+        self.encoder_dim = encoder_dim
+        self.encoder_rates = encoder_rates
+        self.decoder_dim = decoder_dim
+        self.decoder_rates = decoder_rates
+        self.sample_rate = sample_rate
+
+        if latent_dim is None:
+            latent_dim = encoder_dim * (2 ** len(encoder_rates))
+
+        self.latent_dim = latent_dim
+
+        self.hop_length = np.prod(encoder_rates)
+        self.encoder = Encoder(encoder_dim, encoder_rates, latent_dim)
+
+        self.n_codebooks = n_codebooks
+        self.codebook_size = codebook_size
+        self.codebook_dim = codebook_dim
+        self.quantizer = ResidualVectorQuantize(
+            input_dim=latent_dim,
+            n_codebooks=n_codebooks,
+            codebook_size=codebook_size,
+            codebook_dim=codebook_dim,
+            quantizer_dropout=quantizer_dropout,
+        )
+
+        self.decoder = Decoder(
+            latent_dim,
+            decoder_dim,
+            decoder_rates,
+        )
+        self.sample_rate = sample_rate
+        self.apply(init_weights)
+
+        self.delay = self.get_delay()
+
+    def preprocess(self, audio_data, sample_rate):
+        if sample_rate is None:
+            sample_rate = self.sample_rate
+        assert sample_rate == self.sample_rate
+
+        length = audio_data.shape[-1]
+        right_pad = math.ceil(length / self.hop_length) * self.hop_length - length
+        audio_data = nn.functional.pad(audio_data, (0, right_pad))
+
+        return audio_data
+
+    def encode(
+        self,
+        audio_data: torch.Tensor,
+        n_quantizers: int = None,
+    ):
+        """Encode given audio data and return quantized latent codes
+
+        Parameters
+        ----------
+        audio_data : Tensor[B x 1 x T]
+            Audio data to encode
+        n_quantizers : int, optional
+            Number of quantizers to use, by default None
+            If None, all quantizers are used.
+
+        Returns
+        -------
+        dict
+            A dictionary with the following keys:
+            "z" : Tensor[B x D x T]
+                Quantized continuous representation of input
+            "codes" : Tensor[B x N x T]
+                Codebook indices for each codebook
+                (quantized discrete representation of input)
+            "latents" : Tensor[B x N*D x T]
+                Projected latents (continuous representation of input before quantization)
+            "vq/commitment_loss" : Tensor[1]
+                Commitment loss to train encoder to predict vectors closer to codebook
+                entries
+            "vq/codebook_loss" : Tensor[1]
+                Codebook loss to update the codebook
+            "length" : int
+                Number of samples in input audio
+        """
+        z = self.encoder(audio_data)
+        z, codes, latents, commitment_loss, codebook_loss = self.quantizer(z, n_quantizers)
+        return z, codes, latents, commitment_loss, codebook_loss
+
+    def decode(self, z: torch.Tensor):
+        """Decode given latent codes and return audio data
+
+        Parameters
+        ----------
+        z : Tensor[B x D x T]
+            Quantized continuous representation of input
+        length : int, optional
+            Number of samples in output audio, by default None
+
+        Returns
+        -------
+        dict
+            A dictionary with the following keys:
+            "audio" : Tensor[B x 1 x length]
+                Decoded audio data.
+        """
+        return self.decoder(z)
+
+    def forward(
+        self,
+        audio_data: torch.Tensor,
+        sample_rate: int = None,
+        n_quantizers: int = None,
+    ):
+        """Model forward pass
+
+        Parameters
+        ----------
+        audio_data : Tensor[B x 1 x T]
+            Audio data to encode
+        sample_rate : int, optional
+            Sample rate of audio data in Hz, by default None
+            If None, defaults to `self.sample_rate`
+        n_quantizers : int, optional
+            Number of quantizers to use, by default None.
+            If None, all quantizers are used.
+
+        Returns
+        -------
+        dict
+            A dictionary with the following keys:
+            "z" : Tensor[B x D x T]
+                Quantized continuous representation of input
+            "codes" : Tensor[B x N x T]
+                Codebook indices for each codebook
+                (quantized discrete representation of input)
+            "latents" : Tensor[B x N*D x T]
+                Projected latents (continuous representation of input before quantization)
+            "vq/commitment_loss" : Tensor[1]
+                Commitment loss to train encoder to predict vectors closer to codebook
+                entries
+            "vq/codebook_loss" : Tensor[1]
+                Codebook loss to update the codebook
+            "length" : int
+                Number of samples in input audio
+            "audio" : Tensor[B x 1 x length]
+                Decoded audio data.
+        """
+        length = audio_data.shape[-1]
+        audio_data = self.preprocess(audio_data, sample_rate)
+        z, codes, latents, commitment_loss, codebook_loss = self.encode(audio_data, n_quantizers)
+
+        x = self.decode(z)
+        return {
+            "audio": x[..., :length],
+            "z": z,
+            "codes": codes,
+            "latents": latents,
+            "vq/commitment_loss": commitment_loss,
+            "vq/codebook_loss": codebook_loss,
+        }
+
+
+if __name__ == "__main__":
+    import numpy as np
+    from functools import partial
+
+    model = DAC().to("cpu")
+
+    for n, m in model.named_modules():
+        o = m.extra_repr()
+        p = sum([np.prod(p.size()) for p in m.parameters()])
+        fn = lambda o, p: o + f" {p / 1e6:<.3f}M params."
+        setattr(m, "extra_repr", partial(fn, o=o, p=p))
+    print(model)
+    print("Total # of params: ", sum([np.prod(p.size()) for p in model.parameters()]))
+
+    length = 88200 * 2
+    x = torch.randn(1, 1, length).to(model.device)
+    x.requires_grad_(True)
+    x.retain_grad()
+
+    # Make a forward pass
+    out = model(x)["audio"]
+    print("Input shape:", x.shape)
+    print("Output shape:", out.shape)
+
+    # Create gradient variable
+    grad = torch.zeros_like(out)
+    grad[:, :, grad.shape[-1] // 2] = 1
+
+    # Make a backward pass
+    out.backward(grad)
+
+    # Check non-zero values
+    gradmap = x.grad.squeeze(0)
+    gradmap = (gradmap != 0).sum(0)  # sum across features
+    rf = (gradmap != 0).sum()
+
+    print(f"Receptive field: {rf.item()}")
+
+    x = AudioSignal(torch.randn(1, 1, 44100 * 60), 44100)
+    model.decompress(model.compress(x, verbose=True), verbose=True)

train-higgs-audio/boson_multimodal/audio_processing/descriptaudiocodec/dac/nn/layers.py

@@ -0,0 +1,33 @@
+import numpy as np
+import torch
+import torch.nn as nn
+import torch.nn.functional as F
+from einops import rearrange
+from torch.nn.utils import weight_norm
+
+
+def WNConv1d(*args, **kwargs):
+    return weight_norm(nn.Conv1d(*args, **kwargs))
+
+
+def WNConvTranspose1d(*args, **kwargs):
+    return weight_norm(nn.ConvTranspose1d(*args, **kwargs))
+
+
+# Scripting this brings model speed up 1.4x
+@torch.jit.script
+def snake(x, alpha):
+    shape = x.shape
+    x = x.reshape(shape[0], shape[1], -1)
+    x = x + (alpha + 1e-9).reciprocal() * torch.sin(alpha * x).pow(2)
+    x = x.reshape(shape)
+    return x
+
+
+class Snake1d(nn.Module):
+    def __init__(self, channels):
+        super().__init__()
+        self.alpha = nn.Parameter(torch.ones(1, channels, 1))
+
+    def forward(self, x):
+        return snake(x, self.alpha)

train-higgs-audio/boson_multimodal/audio_processing/descriptaudiocodec/dac/nn/quantize.py

@@ -0,0 +1,251 @@
+from typing import Union
+
+import numpy as np
+import torch
+import torch.nn as nn
+import torch.nn.functional as F
+from einops import rearrange
+from torch.nn.utils import weight_norm
+
+from dac.nn.layers import WNConv1d
+
+
+class VectorQuantize(nn.Module):
+    """
+    Implementation of VQ similar to Karpathy's repo:
+    https://github.com/karpathy/deep-vector-quantization
+    Additionally uses following tricks from Improved VQGAN
+    (https://arxiv.org/pdf/2110.04627.pdf):
+        1. Factorized codes: Perform nearest neighbor lookup in low-dimensional space
+            for improved codebook usage
+        2. l2-normalized codes: Converts euclidean distance to cosine similarity which
+            improves training stability
+    """
+
+    def __init__(self, input_dim: int, codebook_size: int, codebook_dim: int):
+        super().__init__()
+        self.codebook_size = codebook_size
+        self.codebook_dim = codebook_dim
+
+        self.in_proj = WNConv1d(input_dim, codebook_dim, kernel_size=1)
+        self.out_proj = WNConv1d(codebook_dim, input_dim, kernel_size=1)
+        self.codebook = nn.Embedding(codebook_size, codebook_dim)
+
+    def forward(self, z):
+        """Quantized the input tensor using a fixed codebook and returns
+        the corresponding codebook vectors
+
+        Parameters
+        ----------
+        z : Tensor[B x D x T]
+
+        Returns
+        -------
+        Tensor[B x D x T]
+            Quantized continuous representation of input
+        Tensor[1]
+            Commitment loss to train encoder to predict vectors closer to codebook
+            entries
+        Tensor[1]
+            Codebook loss to update the codebook
+        Tensor[B x T]
+            Codebook indices (quantized discrete representation of input)
+        Tensor[B x D x T]
+            Projected latents (continuous representation of input before quantization)
+        """
+
+        # Factorized codes (ViT-VQGAN) Project input into low-dimensional space
+        z_e = self.in_proj(z)  # z_e : (B x D x T)
+        z_q, indices = self.decode_latents(z_e)
+
+        commitment_loss = F.mse_loss(z_e, z_q.detach(), reduction="none").mean([1, 2])
+        codebook_loss = F.mse_loss(z_q, z_e.detach(), reduction="none").mean([1, 2])
+
+        z_q = z_e + (z_q - z_e).detach()  # noop in forward pass, straight-through gradient estimator in backward pass
+
+        z_q = self.out_proj(z_q)
+
+        return z_q, commitment_loss, codebook_loss, indices, z_e
+
+    def embed_code(self, embed_id):
+        return F.embedding(embed_id, self.codebook.weight)
+
+    def decode_code(self, embed_id):
+        return self.embed_code(embed_id).transpose(1, 2)
+
+    def decode_latents(self, latents):
+        encodings = rearrange(latents, "b d t -> (b t) d")
+        codebook = self.codebook.weight  # codebook: (N x D)
+
+        # L2 normalize encodings and codebook (ViT-VQGAN)
+        encodings = F.normalize(encodings)
+        codebook = F.normalize(codebook)
+
+        # Compute euclidean distance with codebook
+        dist = (
+            encodings.pow(2).sum(1, keepdim=True)
+            - 2 * encodings @ codebook.t()
+            + codebook.pow(2).sum(1, keepdim=True).t()
+        )
+        indices = rearrange((-dist).max(1)[1], "(b t) -> b t", b=latents.size(0))
+        z_q = self.decode_code(indices)
+        return z_q, indices
+
+
+class ResidualVectorQuantize(nn.Module):
+    """
+    Introduced in SoundStream: An end2end neural audio codec
+    https://arxiv.org/abs/2107.03312
+    """
+
+    def __init__(
+        self,
+        input_dim: int = 512,
+        n_codebooks: int = 9,
+        codebook_size: int = 1024,
+        codebook_dim: Union[int, list] = 8,
+        quantizer_dropout: float = 0.0,
+    ):
+        super().__init__()
+        if isinstance(codebook_dim, int):
+            codebook_dim = [codebook_dim for _ in range(n_codebooks)]
+
+        self.n_codebooks = n_codebooks
+        self.codebook_dim = codebook_dim
+        self.codebook_size = codebook_size
+
+        self.quantizers = nn.ModuleList(
+            [VectorQuantize(input_dim, codebook_size, codebook_dim[i]) for i in range(n_codebooks)]
+        )
+        self.quantizer_dropout = quantizer_dropout
+
+    def forward(self, z, n_quantizers: int = None):
+        """Quantized the input tensor using a fixed set of `n` codebooks and returns
+        the corresponding codebook vectors
+        Parameters
+        ----------
+        z : Tensor[B x D x T]
+        n_quantizers : int, optional
+            No. of quantizers to use
+            (n_quantizers < self.n_codebooks ex: for quantizer dropout)
+            Note: if `self.quantizer_dropout` is True, this argument is ignored
+                when in training mode, and a random number of quantizers is used.
+        Returns
+        -------
+        dict
+            A dictionary with the following keys:
+
+            "z" : Tensor[B x D x T]
+                Quantized continuous representation of input
+            "codes" : Tensor[B x N x T]
+                Codebook indices for each codebook
+                (quantized discrete representation of input)
+            "latents" : Tensor[B x N*D x T]
+                Projected latents (continuous representation of input before quantization)
+            "vq/commitment_loss" : Tensor[1]
+                Commitment loss to train encoder to predict vectors closer to codebook
+                entries
+            "vq/codebook_loss" : Tensor[1]
+                Codebook loss to update the codebook
+        """
+        z_q = 0
+        residual = z
+        commitment_loss = 0
+        codebook_loss = 0
+
+        codebook_indices = []
+        latents = []
+
+        if n_quantizers is None:
+            n_quantizers = self.n_codebooks
+        if self.training:
+            n_quantizers = torch.ones((z.shape[0],)) * self.n_codebooks + 1
+            dropout = torch.randint(1, self.n_codebooks + 1, (z.shape[0],))
+            n_dropout = int(z.shape[0] * self.quantizer_dropout)
+            n_quantizers[:n_dropout] = dropout[:n_dropout]
+            n_quantizers = n_quantizers.to(z.device)
+
+        for i, quantizer in enumerate(self.quantizers):
+            if self.training is False and i >= n_quantizers:
+                break
+
+            z_q_i, commitment_loss_i, codebook_loss_i, indices_i, z_e_i = quantizer(residual)
+
+            # Create mask to apply quantizer dropout
+            mask = torch.full((z.shape[0],), fill_value=i, device=z.device) < n_quantizers
+            z_q = z_q + z_q_i * mask[:, None, None]
+            residual = residual - z_q_i
+
+            # Sum losses
+            commitment_loss += (commitment_loss_i * mask).mean()
+            codebook_loss += (codebook_loss_i * mask).mean()
+
+            codebook_indices.append(indices_i)
+            latents.append(z_e_i)
+
+        codes = torch.stack(codebook_indices, dim=1)
+        latents = torch.cat(latents, dim=1)
+
+        return z_q, codes, latents, commitment_loss, codebook_loss
+
+    def from_codes(self, codes: torch.Tensor):
+        """Given the quantized codes, reconstruct the continuous representation
+        Parameters
+        ----------
+        codes : Tensor[B x N x T]
+            Quantized discrete representation of input
+        Returns
+        -------
+        Tensor[B x D x T]
+            Quantized continuous representation of input
+        """
+        z_q = 0.0
+        z_p = []
+        n_codebooks = codes.shape[1]
+        for i in range(n_codebooks):
+            z_p_i = self.quantizers[i].decode_code(codes[:, i, :])
+            z_p.append(z_p_i)
+
+            z_q_i = self.quantizers[i].out_proj(z_p_i)
+            z_q = z_q + z_q_i
+        return z_q, torch.cat(z_p, dim=1), codes
+
+    def from_latents(self, latents: torch.Tensor):
+        """Given the unquantized latents, reconstruct the
+        continuous representation after quantization.
+
+        Parameters
+        ----------
+        latents : Tensor[B x N x T]
+            Continuous representation of input after projection
+
+        Returns
+        -------
+        Tensor[B x D x T]
+            Quantized representation of full-projected space
+        Tensor[B x D x T]
+            Quantized representation of latent space
+        """
+        z_q = 0
+        z_p = []
+        codes = []
+        dims = np.cumsum([0] + [q.codebook_dim for q in self.quantizers])
+
+        n_codebooks = np.where(dims <= latents.shape[1])[0].max(axis=0, keepdims=True)[0]
+        for i in range(n_codebooks):
+            j, k = dims[i], dims[i + 1]
+            z_p_i, codes_i = self.quantizers[i].decode_latents(latents[:, j:k, :])
+            z_p.append(z_p_i)
+            codes.append(codes_i)
+
+            z_q_i = self.quantizers[i].out_proj(z_p_i)
+            z_q = z_q + z_q_i
+
+        return z_q, torch.cat(z_p, dim=1), torch.stack(codes, dim=1)
+
+
+if __name__ == "__main__":
+    rvq = ResidualVectorQuantize(quantizer_dropout=True)
+    x = torch.randn(16, 512, 80)
+    y = rvq(x)
+    print(y["latents"].shape)

train-higgs-audio/boson_multimodal/audio_processing/higgs_audio_tokenizer.py

@@ -0,0 +1,327 @@
+# Based on code from: https://github.com/zhenye234/xcodec
+# Licensed under MIT License
+# Modifications by BosonAI
+
+import math
+import os
+import torch
+import torch.nn as nn
+import torch.nn.functional as F
+from typing import Optional, Union, Sequence
+import numpy as np
+from transformers import AutoModel
+import torchaudio
+import json
+import librosa
+from huggingface_hub import snapshot_download
+
+from vector_quantize_pytorch import ResidualFSQ
+from .descriptaudiocodec.dac.model import dac as dac2
+from .quantization.vq import ResidualVectorQuantizer
+from .semantic_module import Encoder, Decoder
+
+
+class EncodedResult:
+    def __init__(self, audio_codes):
+        self.audio_codes = audio_codes
+
+
+class HiggsAudioFeatureExtractor(nn.Module):
+    def __init__(self, sampling_rate=16000):
+        super().__init__()
+        self.sampling_rate = sampling_rate
+
+    def forward(self, raw_audio, sampling_rate=16000, return_tensors="pt"):
+        # Convert from librosa to torch
+        audio_signal = torch.tensor(raw_audio)
+        audio_signal = audio_signal.unsqueeze(0)
+        if len(audio_signal.shape) < 3:
+            audio_signal = audio_signal.unsqueeze(0)
+        return {"input_values": audio_signal}
+
+
+class HiggsAudioTokenizer(nn.Module):
+    def __init__(
+        self,
+        n_filters: int = 32,
+        D: int = 128,
+        target_bandwidths: Sequence[Union[int, float]] = [1, 1.5, 2, 4, 6],
+        ratios: Sequence[int] = [8, 5, 4, 2],  #  downsampling by 320
+        sample_rate: int = 16000,
+        bins: int = 1024,
+        n_q: int = 8,
+        codebook_dim: int = None,
+        normalize: bool = False,
+        causal: bool = False,
+        semantic_techer: str = "hubert_base_general",
+        last_layer_semantic: bool = True,
+        merge_mode: str = "concat",
+        downsample_mode: str = "step_down",
+        semantic_mode: str = "classic",
+        vq_scale: int = 1,
+        semantic_sample_rate: int = None,
+        device: str = "cuda",
+    ):
+        super().__init__()
+        self.hop_length = np.prod(ratios)
+        self.semantic_techer = semantic_techer
+
+        self.frame_rate = math.ceil(sample_rate / np.prod(ratios))  # 50 Hz
+
+        self.target_bandwidths = target_bandwidths
+        self.n_q = n_q
+        self.sample_rate = sample_rate
+        self.encoder = dac2.Encoder(64, ratios, D)
+
+        self.decoder_2 = dac2.Decoder(D, 1024, ratios)
+        self.last_layer_semantic = last_layer_semantic
+        self.device = device
+        if semantic_techer == "hubert_base":
+            self.semantic_model = AutoModel.from_pretrained("facebook/hubert-base-ls960")
+            self.semantic_sample_rate = 16000
+            self.semantic_dim = 768
+            self.encoder_semantic_dim = 768
+
+        elif semantic_techer == "wavlm_base_plus":
+            self.semantic_model = AutoModel.from_pretrained("microsoft/wavlm-base-plus")
+            self.semantic_sample_rate = 16000
+            self.semantic_dim = 768
+            self.encoder_semantic_dim = 768
+
+        elif semantic_techer == "hubert_base_general":
+            self.semantic_model = AutoModel.from_pretrained("bosonai/hubert_base", trust_remote_code=True)
+            self.semantic_sample_rate = 16000
+            self.semantic_dim = 768
+            self.encoder_semantic_dim = 768
+
+        # Overwrite semantic model sr to ensure semantic_downsample_factor is an integer
+        if semantic_sample_rate is not None:
+            self.semantic_sample_rate = semantic_sample_rate
+
+        self.semantic_model.eval()
+
+        # make the semantic model parameters do not need gradient
+        for param in self.semantic_model.parameters():
+            param.requires_grad = False
+
+        self.semantic_downsample_factor = int(self.hop_length / (self.sample_rate / self.semantic_sample_rate) / 320)
+
+        self.quantizer_dim = int((D + self.encoder_semantic_dim) // vq_scale)
+        self.encoder_semantic = Encoder(input_channels=self.semantic_dim, encode_channels=self.encoder_semantic_dim)
+        self.decoder_semantic = Decoder(
+            code_dim=self.encoder_semantic_dim, output_channels=self.semantic_dim, decode_channels=self.semantic_dim
+        )
+
+        # out_D=D+768
+        if isinstance(bins, int):  # RVQ
+            self.quantizer = ResidualVectorQuantizer(
+                dimension=self.quantizer_dim, codebook_dim=codebook_dim, n_q=n_q, bins=bins
+            )
+            self.quantizer_type = "RVQ"
+        else:  # RFSQ
+            self.quantizer = ResidualFSQ(dim=self.quantizer_dim, levels=bins, num_quantizers=n_q)
+            self.quantizer_type = "RFSQ"
+
+        self.fc_prior = nn.Linear(D + self.encoder_semantic_dim, self.quantizer_dim)
+        self.fc_post1 = nn.Linear(self.quantizer_dim, self.encoder_semantic_dim)
+        self.fc_post2 = nn.Linear(self.quantizer_dim, D)
+
+        self.downsample_mode = downsample_mode
+        if downsample_mode == "avg":
+            self.semantic_pooling = nn.AvgPool1d(
+                kernel_size=self.semantic_downsample_factor, stride=self.semantic_downsample_factor
+            )
+
+        self.audio_tokenizer_feature_extractor = HiggsAudioFeatureExtractor(sampling_rate=self.sample_rate)
+
+    @property
+    def tps(self):
+        return self.frame_rate
+
+    @property
+    def sampling_rate(self):
+        return self.sample_rate
+
+    @property
+    def num_codebooks(self):
+        return self.n_q
+
+    @property
+    def codebook_size(self):
+        return self.quantizer_dim
+
+    def get_last_layer(self):
+        return self.decoder.layers[-1].weight
+
+    def calculate_rec_loss(self, rec, target):
+        target = target / target.norm(dim=-1, keepdim=True)
+        rec = rec / rec.norm(dim=-1, keepdim=True)
+        rec_loss = (1 - (target * rec).sum(-1)).mean()
+
+        return rec_loss
+
+    @torch.no_grad()
+    def get_regress_target(self, x):
+        x = torchaudio.functional.resample(x, self.sample_rate, self.semantic_sample_rate)
+
+        if (
+            self.semantic_techer == "hubert_base"
+            or self.semantic_techer == "hubert_base_general"
+            or self.semantic_techer == "wavlm_base_plus"
+        ):
+            x = x[:, 0, :]
+            x = F.pad(x, (160, 160))
+            target = self.semantic_model(x, output_hidden_states=True).hidden_states
+            target = torch.stack(target, dim=1)  # .transpose(-1, -2)#.flatten(start_dim=1, end_dim=2)
+
+            # average for all layers
+            target = target.mean(1)
+            # target = target[9]
+            # if self.hop_length > 320:
+            #     target = self.semantic_pooling(target.transpose(1, 2)).transpose(1, 2)
+
+        elif self.semantic_techer == "w2v_bert2":
+            target = self.semantic_model(x)
+
+        elif self.semantic_techer.startswith("whisper"):
+            if self.last_layer_semantic:
+                target = self.semantic_model(x, avg_layers=False)
+            else:
+                target = self.semantic_model(x, avg_layers=True)
+
+        elif self.semantic_techer.startswith("mert_music"):
+            if self.last_layer_semantic:
+                target = self.semantic_model(x, avg_layers=False)
+            else:
+                target = self.semantic_model(x, avg_layers=True)
+
+        elif self.semantic_techer.startswith("qwen_audio_omni"):
+            target = self.semantic_model(x)
+
+        if self.downsample_mode == "step_down":
+            if self.semantic_downsample_factor > 1:
+                target = target[:, :: self.semantic_downsample_factor, :]
+
+        elif self.downsample_mode == "avg":
+            target = self.semantic_pooling(target.transpose(1, 2)).transpose(1, 2)
+        return target
+
+    def forward(self, x: torch.Tensor, bw: int):
+        e_semantic_input = self.get_regress_target(x).detach()
+
+        e_semantic = self.encoder_semantic(e_semantic_input.transpose(1, 2))
+        e_acoustic = self.encoder(x)
+
+        e = torch.cat([e_acoustic, e_semantic], dim=1)
+
+        e = self.fc_prior(e.transpose(1, 2))
+
+        if self.quantizer_type == "RVQ":
+            e = e.transpose(1, 2)
+            quantized, codes, bandwidth, commit_loss = self.quantizer(e, self.frame_rate, bw)
+            quantized = quantized.transpose(1, 2)
+        else:
+            quantized, codes = self.quantizer(e)
+            commit_loss = torch.tensor(0.0)
+
+        quantized_semantic = self.fc_post1(quantized).transpose(1, 2)
+        quantized_acoustic = self.fc_post2(quantized).transpose(1, 2)
+
+        o = self.decoder_2(quantized_acoustic)
+
+        o_semantic = self.decoder_semantic(quantized_semantic)
+        semantic_recon_loss = F.mse_loss(e_semantic_input.transpose(1, 2).detach(), o_semantic)
+
+        return o, commit_loss, semantic_recon_loss, None
+
+    def encode(self, audio_path_or_wv, sr=None, loudness_normalize=False, loudness_threshold=-23.0):
+        if isinstance(audio_path_or_wv, str):
+            wv, sr = librosa.load(audio_path_or_wv, mono=True, sr=None)
+        else:
+            wv = audio_path_or_wv
+            assert sr is not None
+        if loudness_normalize:
+            import pyloudnorm as pyln
+
+            meter = pyln.Meter(sr)
+            l = meter.integrated_loudness(wv)
+            wv = pyln.normalize.loudness(wv, l, loudness_threshold)
+        if sr != self.sampling_rate:
+            wv = librosa.resample(wv, orig_sr=sr, target_sr=self.sampling_rate)
+        if self.audio_tokenizer_feature_extractor is not None:
+            inputs = self.audio_tokenizer_feature_extractor(
+                raw_audio=wv, sampling_rate=self.audio_tokenizer_feature_extractor.sampling_rate, return_tensors="pt"
+            )
+            input_values = inputs["input_values"].to(self.device)
+        else:
+            input_values = torch.from_numpy(wv).float().unsqueeze(0)
+        with torch.no_grad():
+            encoder_outputs = self._xcodec_encode(input_values)
+            vq_code = encoder_outputs.audio_codes[0]
+        return vq_code
+
+    def _xcodec_encode(self, x: torch.Tensor, target_bw: Optional[int] = None) -> torch.Tensor:
+        bw = target_bw
+
+        e_semantic_input = self.get_regress_target(x).detach()
+
+        e_semantic = self.encoder_semantic(e_semantic_input.transpose(1, 2))
+        e_acoustic = self.encoder(x)
+
+        if e_acoustic.shape[2] != e_semantic.shape[2]:
+            pad_size = 160 * self.semantic_downsample_factor
+            e_acoustic = self.encoder(F.pad(x[:, 0, :], (pad_size, pad_size)).unsqueeze(0))
+
+        if e_acoustic.shape[2] != e_semantic.shape[2]:
+            if e_acoustic.shape[2] > e_semantic.shape[2]:
+                e_acoustic = e_acoustic[:, :, : e_semantic.shape[2]]
+            else:
+                e_semantic = e_semantic[:, :, : e_acoustic.shape[2]]
+
+        e = torch.cat([e_acoustic, e_semantic], dim=1)
+
+        e = self.fc_prior(e.transpose(1, 2))
+
+        if self.quantizer_type == "RVQ":
+            e = e.transpose(1, 2)
+            quantized, codes, bandwidth, commit_loss = self.quantizer(e, self.frame_rate, bw)
+            codes = codes.permute(1, 0, 2)
+        else:
+            quantized, codes = self.quantizer(e)
+            codes = codes.permute(0, 2, 1)
+
+        # return codes
+        return EncodedResult(codes)
+
+    def decode(self, vq_code: torch.Tensor) -> torch.Tensor:
+        if self.quantizer_type == "RVQ":
+            vq_code = vq_code.permute(1, 0, 2)
+            quantized = self.quantizer.decode(vq_code)
+            quantized = quantized.transpose(1, 2)
+        else:
+            vq_code = vq_code.permute(0, 2, 1)
+            quantized = self.quantizer.get_output_from_indices(vq_code)
+        quantized_acoustic = self.fc_post2(quantized).transpose(1, 2)
+
+        o = self.decoder_2(quantized_acoustic)
+        return o.cuda().numpy()
+
+
+def load_higgs_audio_tokenizer(tokenizer_name_or_path, device="cuda"):
+    is_local = os.path.exists(tokenizer_name_or_path)
+    if not is_local:
+        tokenizer_path = snapshot_download(tokenizer_name_or_path)
+    else:
+        tokenizer_path = tokenizer_name_or_path
+    config_path = os.path.join(tokenizer_path, "config.json")
+    model_path = os.path.join(tokenizer_path, "model.pth")
+    config = json.load(open(config_path))
+    model = HiggsAudioTokenizer(
+        **config,
+        device=device,
+    )
+    parameter_dict = torch.load(model_path, map_location=device)
+    model.load_state_dict(parameter_dict, strict=False)
+    model.to(device)
+    model.eval()
+    return model

train-higgs-audio/boson_multimodal/audio_processing/quantization/__init__.py

@@ -0,0 +1,8 @@
+# Copyright (c) Meta Platforms, Inc. and affiliates.
+# All rights reserved.
+#
+# This source code is licensed under the license found in the
+# LICENSE file in the root directory of this source tree.
+
+# flake8: noqa
+from .vq import QuantizedResult, ResidualVectorQuantizer

train-higgs-audio/boson_multimodal/audio_processing/quantization/ac.py

@@ -0,0 +1,292 @@
+# Copyright (c) Meta Platforms, Inc. and affiliates.
+# All rights reserved.
+#
+# This source code is licensed under the license found in the
+# LICENSE file in the root directory of this source tree.
+
+"""Arithmetic coder."""
+
+import io
+import math
+import random
+import typing as tp
+import torch
+
+from ..binary import BitPacker, BitUnpacker
+
+
+def build_stable_quantized_cdf(
+    pdf: torch.Tensor, total_range_bits: int, roundoff: float = 1e-8, min_range: int = 2, check: bool = True
+) -> torch.Tensor:
+    """Turn the given PDF into a quantized CDF that splits
+    [0, 2 ** self.total_range_bits - 1] into chunks of size roughly proportional
+    to the PDF.
+
+    Args:
+        pdf (torch.Tensor): probability distribution, shape should be `[N]`.
+        total_range_bits (int): see `ArithmeticCoder`, the typical range we expect
+            during the coding process is `[0, 2 ** total_range_bits - 1]`.
+        roundoff (float): will round the pdf up to that level to remove difference coming
+        from e.g. evaluating the Language Model on different architectures.
+        min_range (int): minimum range width. Should always be at least 2 for numerical
+            stability. Use this to avoid pathological behavior is a value
+            that is expected to be rare actually happens in real life.
+        check (bool): if True, checks that nothing bad happened, can be deactivated for speed.
+    """
+    pdf = pdf.detach()
+    if roundoff:
+        pdf = (pdf / roundoff).floor() * roundoff
+    # interpolate with uniform distribution to achieve desired minimum probability.
+    total_range = 2**total_range_bits
+    cardinality = len(pdf)
+    alpha = min_range * cardinality / total_range
+    assert alpha <= 1, "you must reduce min_range"
+    ranges = (((1 - alpha) * total_range) * pdf).floor().long()
+    ranges += min_range
+    quantized_cdf = torch.cumsum(ranges, dim=-1)
+    if min_range < 2:
+        raise ValueError("min_range must be at least 2.")
+    if check:
+        assert quantized_cdf[-1] <= 2**total_range_bits, quantized_cdf[-1]
+        if ((quantized_cdf[1:] - quantized_cdf[:-1]) < min_range).any() or quantized_cdf[0] < min_range:
+            raise ValueError("You must increase your total_range_bits.")
+    return quantized_cdf
+
+
+class ArithmeticCoder:
+    """ArithmeticCoder,
+    Let us take a distribution `p` over `N` symbols, and assume we have a stream
+    of random variables `s_t` sampled from `p`. Let us assume that we have a budget
+    of `B` bits that we can afford to write on device. There are `2**B` possible numbers,
+    corresponding to the range `[0, 2 ** B - 1]`. We can map each of those number to a single
+    sequence `(s_t)` by doing the following:
+
+    1) Initialize the current range to` [0 ** 2 B - 1]`.
+    2) For each time step t, split the current range into contiguous chunks,
+        one for each possible outcome, with size roughly proportional to `p`.
+        For instance, if `p = [0.75, 0.25]`, and the range is `[0, 3]`, the chunks
+        would be `{[0, 2], [3, 3]}`.
+    3) Select the chunk corresponding to `s_t`, and replace the current range with this.
+    4) When done encoding all the values, just select any value remaining in the range.
+
+    You will notice that this procedure can fail: for instance if at any point in time
+    the range is smaller than `N`, then we can no longer assign a non-empty chunk to each
+    possible outcome. Intuitively, the more likely a value is, the less the range width
+    will reduce, and the longer we can go on encoding values. This makes sense: for any efficient
+    coding scheme, likely outcomes would take less bits, and more of them can be coded
+    with a fixed budget.
+
+    In practice, we do not know `B` ahead of time, but we have a way to inject new bits
+    when the current range decreases below a given limit (given by `total_range_bits`), without
+    having to redo all the computations. If we encode mostly likely values, we will seldom
+    need to inject new bits, but a single rare value can deplete our stock of entropy!
+
+    In this explanation, we assumed that the distribution `p` was constant. In fact, the present
+    code works for any sequence `(p_t)` possibly different for each timestep.
+    We also assume that `s_t ~ p_t`, but that doesn't need to be true, although the smaller
+    the KL between the true distribution and `p_t`, the most efficient the coding will be.
+
+    Args:
+        fo (IO[bytes]): file-like object to which the bytes will be written to.
+        total_range_bits (int): the range `M` described above is `2 ** total_range_bits.
+            Any time the current range width fall under this limit, new bits will
+            be injected to rescale the initial range.
+    """
+
+    def __init__(self, fo: tp.IO[bytes], total_range_bits: int = 24):
+        assert total_range_bits <= 30
+        self.total_range_bits = total_range_bits
+        self.packer = BitPacker(bits=1, fo=fo)  # we push single bits at a time.
+        self.low: int = 0
+        self.high: int = 0
+        self.max_bit: int = -1
+        self._dbg: tp.List[tp.Any] = []
+        self._dbg2: tp.List[tp.Any] = []
+
+    @property
+    def delta(self) -> int:
+        """Return the current range width."""
+        return self.high - self.low + 1
+
+    def _flush_common_prefix(self):
+        # If self.low and self.high start with the sames bits,
+        # those won't change anymore as we always just increase the range
+        # by powers of 2, and we can flush them out to the bit stream.
+        assert self.high >= self.low, (self.low, self.high)
+        assert self.high < 2 ** (self.max_bit + 1)
+        while self.max_bit >= 0:
+            b1 = self.low >> self.max_bit
+            b2 = self.high >> self.max_bit
+            if b1 == b2:
+                self.low -= b1 << self.max_bit
+                self.high -= b1 << self.max_bit
+                assert self.high >= self.low, (self.high, self.low, self.max_bit)
+                assert self.low >= 0
+                self.max_bit -= 1
+                self.packer.push(b1)
+            else:
+                break
+
+    def push(self, symbol: int, quantized_cdf: torch.Tensor):
+        """Push the given symbol on the stream, flushing out bits
+        if possible.
+
+        Args:
+            symbol (int): symbol to encode with the AC.
+            quantized_cdf (torch.Tensor): use `build_stable_quantized_cdf`
+                to build this from your pdf estimate.
+        """
+        while self.delta < 2**self.total_range_bits:
+            self.low *= 2
+            self.high = self.high * 2 + 1
+            self.max_bit += 1
+
+        range_low = 0 if symbol == 0 else quantized_cdf[symbol - 1].item()
+        range_high = quantized_cdf[symbol].item() - 1
+        effective_low = int(math.ceil(range_low * (self.delta / (2**self.total_range_bits))))
+        effective_high = int(math.floor(range_high * (self.delta / (2**self.total_range_bits))))
+        assert self.low <= self.high
+        self.high = self.low + effective_high
+        self.low = self.low + effective_low
+        assert self.low <= self.high, (effective_low, effective_high, range_low, range_high)
+        self._dbg.append((self.low, self.high))
+        self._dbg2.append((self.low, self.high))
+        outs = self._flush_common_prefix()
+        assert self.low <= self.high
+        assert self.max_bit >= -1
+        assert self.max_bit <= 61, self.max_bit
+        return outs
+
+    def flush(self):
+        """Flush the remaining information to the stream."""
+        while self.max_bit >= 0:
+            b1 = (self.low >> self.max_bit) & 1
+            self.packer.push(b1)
+            self.max_bit -= 1
+        self.packer.flush()
+
+
+class ArithmeticDecoder:
+    """ArithmeticDecoder, see `ArithmeticCoder` for a detailed explanation.
+
+    Note that this must be called with **exactly** the same parameters and sequence
+    of quantized cdf as the arithmetic encoder or the wrong values will be decoded.
+
+    If the AC encoder current range is [L, H], with `L` and `H` having the some common
+    prefix (i.e. the same most significant bits), then this prefix will be flushed to the stream.
+    For instances, having read 3 bits `b1 b2 b3`, we know that `[L, H]` is contained inside
+    `[b1 b2 b3 0 ... 0 b1 b3 b3 1 ... 1]`. Now this specific sub-range can only be obtained
+    for a specific sequence of symbols and a binary-search allows us to decode those symbols.
+    At some point, the prefix `b1 b2 b3` will no longer be sufficient to decode new symbols,
+    and we will need to read new bits from the stream and repeat the process.
+
+    """
+
+    def __init__(self, fo: tp.IO[bytes], total_range_bits: int = 24):
+        self.total_range_bits = total_range_bits
+        self.low: int = 0
+        self.high: int = 0
+        self.current: int = 0
+        self.max_bit: int = -1
+        self.unpacker = BitUnpacker(bits=1, fo=fo)  # we pull single bits at a time.
+        # Following is for debugging
+        self._dbg: tp.List[tp.Any] = []
+        self._dbg2: tp.List[tp.Any] = []
+        self._last: tp.Any = None
+
+    @property
+    def delta(self) -> int:
+        return self.high - self.low + 1
+
+    def _flush_common_prefix(self):
+        # Given the current range [L, H], if both have a common prefix,
+        # we know we can remove it from our representation to avoid handling large numbers.
+        while self.max_bit >= 0:
+            b1 = self.low >> self.max_bit
+            b2 = self.high >> self.max_bit
+            if b1 == b2:
+                self.low -= b1 << self.max_bit
+                self.high -= b1 << self.max_bit
+                self.current -= b1 << self.max_bit
+                assert self.high >= self.low
+                assert self.low >= 0
+                self.max_bit -= 1
+            else:
+                break
+
+    def pull(self, quantized_cdf: torch.Tensor) -> tp.Optional[int]:
+        """Pull a symbol, reading as many bits from the stream as required.
+        This returns `None` when the stream has been exhausted.
+
+        Args:
+            quantized_cdf (torch.Tensor): use `build_stable_quantized_cdf`
+                to build this from your pdf estimate. This must be **exatly**
+                the same cdf as the one used at encoding time.
+        """
+        while self.delta < 2**self.total_range_bits:
+            bit = self.unpacker.pull()
+            if bit is None:
+                return None
+            self.low *= 2
+            self.high = self.high * 2 + 1
+            self.current = self.current * 2 + bit
+            self.max_bit += 1
+
+        def bin_search(low_idx: int, high_idx: int):
+            # Binary search is not just for coding interviews :)
+            if high_idx < low_idx:
+                raise RuntimeError("Binary search failed")
+            mid = (low_idx + high_idx) // 2
+            range_low = quantized_cdf[mid - 1].item() if mid > 0 else 0
+            range_high = quantized_cdf[mid].item() - 1
+            effective_low = int(math.ceil(range_low * (self.delta / (2**self.total_range_bits))))
+            effective_high = int(math.floor(range_high * (self.delta / (2**self.total_range_bits))))
+            low = effective_low + self.low
+            high = effective_high + self.low
+            if self.current >= low:
+                if self.current <= high:
+                    return (mid, low, high, self.current)
+                else:
+                    return bin_search(mid + 1, high_idx)
+            else:
+                return bin_search(low_idx, mid - 1)
+
+        self._last = (self.low, self.high, self.current, self.max_bit)
+        sym, self.low, self.high, self.current = bin_search(0, len(quantized_cdf) - 1)
+        self._dbg.append((self.low, self.high, self.current))
+        self._flush_common_prefix()
+        self._dbg2.append((self.low, self.high, self.current))
+
+        return sym
+
+
+def test():
+    torch.manual_seed(1234)
+    random.seed(1234)
+    for _ in range(4):
+        pdfs = []
+        cardinality = random.randrange(4000)
+        steps = random.randrange(100, 500)
+        fo = io.BytesIO()
+        encoder = ArithmeticCoder(fo)
+        symbols = []
+        for step in range(steps):
+            pdf = torch.softmax(torch.randn(cardinality), dim=0)
+            pdfs.append(pdf)
+            q_cdf = build_stable_quantized_cdf(pdf, encoder.total_range_bits)
+            symbol = torch.multinomial(pdf, 1).item()
+            symbols.append(symbol)
+            encoder.push(symbol, q_cdf)
+        encoder.flush()
+
+        fo.seek(0)
+        decoder = ArithmeticDecoder(fo)
+        for idx, (pdf, symbol) in enumerate(zip(pdfs, symbols)):
+            q_cdf = build_stable_quantized_cdf(pdf, encoder.total_range_bits)
+            decoded_symbol = decoder.pull(q_cdf)
+            assert decoded_symbol == symbol, idx
+        assert decoder.pull(torch.zeros(1)) is None
+
+
+if __name__ == "__main__":
+    test()

train-higgs-audio/boson_multimodal/audio_processing/quantization/core_vq.py

@@ -0,0 +1,360 @@
+# Copyright (c) Meta Platforms, Inc. and affiliates.
+# All rights reserved.
+#
+# This source code is licensed under the license found in the
+# LICENSE file in the root directory of this source tree.
+#
+# This implementation is inspired from
+# https://github.com/lucidrains/vector-quantize-pytorch
+# which is released under MIT License. Hereafter, the original license:
+# MIT License
+#
+# Copyright (c) 2020 Phil Wang
+#
+# Permission is hereby granted, free of charge, to any person obtaining a copy
+# of this software and associated documentation files (the "Software"), to deal
+# in the Software without restriction, including without limitation the rights
+# to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+# copies of the Software, and to permit persons to whom the Software is
+# furnished to do so, subject to the following conditions:
+#
+# The above copyright notice and this permission notice shall be included in all
+# copies or substantial portions of the Software.
+#
+# THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+# IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+# FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+# AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+# LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+# OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+# SOFTWARE.
+
+"""Core vector quantization implementation."""
+
+import typing as tp
+
+from einops import rearrange, repeat
+import torch
+from torch import nn
+import torch.nn.functional as F
+
+from xcodec.quantization.distrib import broadcast_tensors, rank
+
+
+def default(val: tp.Any, d: tp.Any) -> tp.Any:
+    return val if val is not None else d
+
+
+def ema_inplace(moving_avg, new, decay: float):
+    moving_avg.data.mul_(decay).add_(new, alpha=(1 - decay))
+
+
+def laplace_smoothing(x, n_categories: int, epsilon: float = 1e-5):
+    return (x + epsilon) / (x.sum() + n_categories * epsilon)
+
+
+def uniform_init(*shape: int):
+    t = torch.empty(shape)
+    nn.init.kaiming_uniform_(t)
+    return t
+
+
+def sample_vectors(samples, num: int):
+    num_samples, device = samples.shape[0], samples.device
+
+    if num_samples >= num:
+        indices = torch.randperm(num_samples, device=device)[:num]
+    else:
+        indices = torch.randint(0, num_samples, (num,), device=device)
+
+    return samples[indices]
+
+
+def kmeans(samples, num_clusters: int, num_iters: int = 10):
+    dim, dtype = samples.shape[-1], samples.dtype
+
+    means = sample_vectors(samples, num_clusters)
+
+    for _ in range(num_iters):
+        diffs = rearrange(samples, "n d -> n () d") - rearrange(means, "c d -> () c d")
+        dists = -(diffs**2).sum(dim=-1)
+
+        buckets = dists.max(dim=-1).indices
+        bins = torch.bincount(buckets, minlength=num_clusters)
+        zero_mask = bins == 0
+        bins_min_clamped = bins.masked_fill(zero_mask, 1)
+
+        new_means = buckets.new_zeros(num_clusters, dim, dtype=dtype)
+        new_means.scatter_add_(0, repeat(buckets, "n -> n d", d=dim), samples)
+        new_means = new_means / bins_min_clamped[..., None]
+
+        means = torch.where(zero_mask[..., None], means, new_means)
+
+    return means, bins
+
+
+class EuclideanCodebook(nn.Module):
+    """Codebook with Euclidean distance.
+    Args:
+        dim (int): Dimension.
+        codebook_size (int): Codebook size.
+        kmeans_init (bool): Whether to use k-means to initialize the codebooks.
+            If set to true, run the k-means algorithm on the first training batch and use
+            the learned centroids as initialization.
+        kmeans_iters (int): Number of iterations used for k-means algorithm at initialization.
+        decay (float): Decay for exponential moving average over the codebooks.
+        epsilon (float): Epsilon value for numerical stability.
+        threshold_ema_dead_code (int): Threshold for dead code expiration. Replace any codes
+            that have an exponential moving average cluster size less than the specified threshold with
+            randomly selected vector from the current batch.
+    """
+
+    def __init__(
+        self,
+        dim: int,
+        codebook_size: int,
+        kmeans_init: int = False,
+        kmeans_iters: int = 10,
+        decay: float = 0.99,
+        epsilon: float = 1e-5,
+        threshold_ema_dead_code: int = 2,
+    ):
+        super().__init__()
+        self.decay = decay
+        init_fn: tp.Union[tp.Callable[..., torch.Tensor], tp.Any] = uniform_init if not kmeans_init else torch.zeros
+        embed = init_fn(codebook_size, dim)
+
+        self.codebook_size = codebook_size
+
+        self.kmeans_iters = kmeans_iters
+        self.epsilon = epsilon
+        self.threshold_ema_dead_code = threshold_ema_dead_code
+
+        self.register_buffer("inited", torch.Tensor([not kmeans_init]))
+        self.register_buffer("cluster_size", torch.zeros(codebook_size))
+        self.register_buffer("embed", embed)
+        self.register_buffer("embed_avg", embed.clone())
+
+    @torch.jit.ignore
+    def init_embed_(self, data):
+        if self.inited:
+            return
+
+        embed, cluster_size = kmeans(data, self.codebook_size, self.kmeans_iters)
+        self.embed.data.copy_(embed)
+        self.embed_avg.data.copy_(embed.clone())
+        self.cluster_size.data.copy_(cluster_size)
+        self.inited.data.copy_(torch.Tensor([True]))
+        # Make sure all buffers across workers are in sync after initialization
+        broadcast_tensors(self.buffers())
+
+    def replace_(self, samples, mask):
+        modified_codebook = torch.where(mask[..., None], sample_vectors(samples, self.codebook_size), self.embed)
+        self.embed.data.copy_(modified_codebook)
+
+    def expire_codes_(self, batch_samples):
+        if self.threshold_ema_dead_code == 0:
+            return
+
+        expired_codes = self.cluster_size < self.threshold_ema_dead_code
+        if not torch.any(expired_codes):
+            return
+
+        batch_samples = rearrange(batch_samples, "... d -> (...) d")
+        self.replace_(batch_samples, mask=expired_codes)
+        broadcast_tensors(self.buffers())
+
+    def preprocess(self, x):
+        x = rearrange(x, "... d -> (...) d")
+        return x
+
+    def quantize(self, x):
+        embed = self.embed.t()
+        dist = -(x.pow(2).sum(1, keepdim=True) - 2 * x @ embed + embed.pow(2).sum(0, keepdim=True))
+        embed_ind = dist.max(dim=-1).indices
+        return embed_ind
+
+    def postprocess_emb(self, embed_ind, shape):
+        return embed_ind.view(*shape[:-1])
+
+    def dequantize(self, embed_ind):
+        quantize = F.embedding(embed_ind, self.embed)  # get embedding based on index
+        return quantize
+
+    def encode(self, x):
+        shape = x.shape
+        # pre-process
+        x = self.preprocess(x)
+        # quantize
+        embed_ind = self.quantize(x)  # get index based on Euclidean distance
+        # post-process
+        embed_ind = self.postprocess_emb(embed_ind, shape)
+        return embed_ind
+
+    def decode(self, embed_ind):
+        quantize = self.dequantize(embed_ind)
+        return quantize
+
+    def forward(self, x):
+        shape, dtype = x.shape, x.dtype
+        x = self.preprocess(x)
+
+        self.init_embed_(x)
+
+        embed_ind = self.quantize(x)
+        embed_onehot = F.one_hot(embed_ind, self.codebook_size).type(dtype)
+        embed_ind = self.postprocess_emb(embed_ind, shape)
+        quantize = self.dequantize(embed_ind)
+
+        if self.training:
+            # We do the expiry of code at that point as buffers are in sync
+            # and all the workers will take the same decision.
+            self.expire_codes_(x)
+            ema_inplace(self.cluster_size, embed_onehot.sum(0), self.decay)
+            embed_sum = x.t() @ embed_onehot
+            ema_inplace(self.embed_avg, embed_sum.t(), self.decay)
+            cluster_size = (
+                laplace_smoothing(self.cluster_size, self.codebook_size, self.epsilon) * self.cluster_size.sum()
+            )
+            embed_normalized = self.embed_avg / cluster_size.unsqueeze(1)
+            self.embed.data.copy_(embed_normalized)
+
+        return quantize, embed_ind
+
+
+class VectorQuantization(nn.Module):
+    """Vector quantization implementation.
+    Currently supports only euclidean distance.
+    Args:
+        dim (int): Dimension
+        codebook_size (int): Codebook size
+        codebook_dim (int): Codebook dimension. If not defined, uses the specified dimension in dim.
+        decay (float): Decay for exponential moving average over the codebooks.
+        epsilon (float): Epsilon value for numerical stability.
+        kmeans_init (bool): Whether to use kmeans to initialize the codebooks.
+        kmeans_iters (int): Number of iterations used for kmeans initialization.
+        threshold_ema_dead_code (int): Threshold for dead code expiration. Replace any codes
+            that have an exponential moving average cluster size less than the specified threshold with
+            randomly selected vector from the current batch.
+        commitment_weight (float): Weight for commitment loss.
+    """
+
+    def __init__(
+        self,
+        dim: int,
+        codebook_size: int,
+        codebook_dim: tp.Optional[int] = None,
+        decay: float = 0.99,
+        epsilon: float = 1e-5,
+        kmeans_init: bool = True,
+        kmeans_iters: int = 50,
+        threshold_ema_dead_code: int = 2,
+        commitment_weight: float = 1.0,
+    ):
+        super().__init__()
+        _codebook_dim: int = default(codebook_dim, dim)
+
+        requires_projection = _codebook_dim != dim
+        self.project_in = nn.Linear(dim, _codebook_dim) if requires_projection else nn.Identity()
+        self.project_out = nn.Linear(_codebook_dim, dim) if requires_projection else nn.Identity()
+
+        self.epsilon = epsilon
+        self.commitment_weight = commitment_weight
+
+        self._codebook = EuclideanCodebook(
+            dim=_codebook_dim,
+            codebook_size=codebook_size,
+            kmeans_init=kmeans_init,
+            kmeans_iters=kmeans_iters,
+            decay=decay,
+            epsilon=epsilon,
+            threshold_ema_dead_code=threshold_ema_dead_code,
+        )
+        self.codebook_size = codebook_size
+
+    @property
+    def codebook(self):
+        return self._codebook.embed
+
+    def encode(self, x):
+        x = rearrange(x, "b d n -> b n d")
+        x = self.project_in(x)
+        embed_in = self._codebook.encode(x)
+        return embed_in
+
+    def decode(self, embed_ind):
+        quantize = self._codebook.decode(embed_ind)
+        quantize = self.project_out(quantize)
+        quantize = rearrange(quantize, "b n d -> b d n")
+        return quantize
+
+    def forward(self, x):
+        device = x.device
+        x = rearrange(x, "b d n -> b n d")
+        x = self.project_in(x)
+
+        quantize, embed_ind = self._codebook(x)
+
+        if self.training:
+            quantize = x + (quantize - x).detach()
+
+        loss = torch.tensor([0.0], device=device, requires_grad=self.training)
+
+        if self.training:
+            if self.commitment_weight > 0:
+                commit_loss = F.mse_loss(quantize.detach(), x)
+                loss = loss + commit_loss * self.commitment_weight
+
+        quantize = self.project_out(quantize)
+        quantize = rearrange(quantize, "b n d -> b d n")
+        return quantize, embed_ind, loss
+
+
+class ResidualVectorQuantization(nn.Module):
+    """Residual vector quantization implementation.
+    Follows Algorithm 1. in https://arxiv.org/pdf/2107.03312.pdf
+    """
+
+    def __init__(self, *, num_quantizers, **kwargs):
+        super().__init__()
+        self.layers = nn.ModuleList([VectorQuantization(**kwargs) for _ in range(num_quantizers)])
+
+    def forward(self, x, n_q: tp.Optional[int] = None):
+        quantized_out = 0.0
+        residual = x
+
+        all_losses = []
+        all_indices = []
+
+        n_q = n_q or len(self.layers)
+
+        for layer in self.layers[:n_q]:
+            quantized, indices, loss = layer(residual)
+            residual = residual - quantized
+            quantized_out = quantized_out + quantized
+
+            all_indices.append(indices)
+            all_losses.append(loss)
+
+        out_losses, out_indices = map(torch.stack, (all_losses, all_indices))
+        return quantized_out, out_indices, out_losses
+
+    def encode(self, x: torch.Tensor, n_q: tp.Optional[int] = None) -> torch.Tensor:
+        residual = x
+        all_indices = []
+        n_q = n_q or len(self.layers)
+        for layer in self.layers[:n_q]:
+            indices = layer.encode(residual)
+            quantized = layer.decode(indices)
+            residual = residual - quantized
+            all_indices.append(indices)
+        out_indices = torch.stack(all_indices)
+        return out_indices
+
+    def decode(self, q_indices: torch.Tensor) -> torch.Tensor:
+        quantized_out = torch.tensor(0.0, device=q_indices.device)
+        for i, indices in enumerate(q_indices):
+            layer = self.layers[i]
+            quantized = layer.decode(indices)
+            quantized_out = quantized_out + quantized
+        return quantized_out

train-higgs-audio/boson_multimodal/audio_processing/quantization/core_vq_lsx_version.py

@@ -0,0 +1,425 @@
+# Copyright (c)
+#
+# This source code is licensed under the license found in the
+# LICENSE file in the root directory of this source tree.
+# This implementation is inspired from
+# https://github.com/rosinality/vq-vae-2-pytorch/blob/master/vqvae.py and
+# https://github.com/clementchadebec/benchmark_VAE/blob/dfa0dcf6c79172df5d27769c09c860c42008baaa/src/pythae/models/vq_vae/vq_vae_utils.py#L81
+#
+# Copyright (c) Meta Platforms, Inc. and affiliates.
+# All rights reserved.
+#
+# This source code is licensed under the license found in the
+# LICENSE file in the root directory of this source tree.
+#
+# This implementation is inspired from
+# https://github.com/lucidrains/vector-quantize-pytorch
+# which is released under MIT License. Hereafter, the original license:
+# MIT License
+#
+# Copyright (c) 2020 Phil Wang
+#
+# Permission is hereby granted, free of charge, to any person obtaining a copy
+# of this software and associated documentation files (the "Software"), to deal
+# in the Software without restriction, including without limitation the rights
+# to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+# copies of the Software, and to permit persons to whom the Software is
+# furnished to do so, subject to the following conditions:
+#
+# The above copyright notice and this permission notice shall be included in all
+# copies or substantial portions of the Software.
+#
+# THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+# IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+# FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+# AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+# LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+# OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+# SOFTWARE.
+
+"""Core vector quantization implementation."""
+
+import typing as tp
+
+from einops import rearrange
+import torch
+from torch import nn
+import torch.nn.functional as F
+import torch.distributed as dist
+
+from .distrib import broadcast_tensors, is_distributed
+from .ddp_utils import SyncFunction
+
+
+def default(val: tp.Any, d: tp.Any) -> tp.Any:
+    return val if val is not None else d
+
+
+def ema_inplace(moving_avg, new, decay: float):
+    moving_avg.data.mul_(decay).add_(new, alpha=(1 - decay))
+
+
+def laplace_smoothing(x, n_categories: int, epsilon: float = 1e-5):
+    return (x + epsilon) / (x.sum() + n_categories * epsilon)
+
+
+def uniform_init(*shape: int):
+    t = torch.empty(shape)
+    nn.init.kaiming_uniform_(t)
+    return t
+
+
+def sample_vectors(samples, num: int):
+    num_samples, device = samples.shape[0], samples.device
+
+    if num_samples >= num:
+        indices = torch.randperm(num_samples, device=device)[:num]
+    else:
+        indices = torch.randint(0, num_samples, (num,), device=device)
+
+    return samples[indices]
+
+
+def kmeans(samples, num_clusters: int, num_iters: int = 10, frames_to_use: int = 10_000, batch_size: int = 64):
+    """
+    Memory-efficient K-means clustering.
+    Args:
+        samples (tensor): shape [N, D]
+        num_clusters (int): number of centroids.
+        num_iters (int): number of iterations.
+        frames_to_use (int): subsample size from total samples.
+        batch_size (int): batch size used in distance computation.
+    Returns:
+        means: [num_clusters, D]
+        bins: [num_clusters] (number of points per cluster)
+    """
+    N, D = samples.shape
+    dtype, device = samples.dtype, samples.device
+
+    if frames_to_use < N:
+        indices = torch.randperm(N, device=device)[:frames_to_use]
+        samples = samples[indices]
+
+    means = sample_vectors(samples, num_clusters)
+
+    for _ in range(num_iters):
+        # Store cluster assignments
+        all_assignments = []
+
+        for i in range(0, samples.shape[0], batch_size):
+            batch = samples[i : i + batch_size]  # [B, D]
+            dists = torch.cdist(batch, means, p=2)  # [B, C]
+            assignments = dists.argmin(dim=1)  # [B]
+            all_assignments.append(assignments)
+
+        buckets = torch.cat(all_assignments, dim=0)  # [N]
+        bins = torch.bincount(buckets, minlength=num_clusters)
+        zero_mask = bins == 0
+        bins_min_clamped = bins.masked_fill(zero_mask, 1)
+
+        # Compute new means
+        new_means = torch.zeros_like(means)
+        for i in range(num_clusters):
+            mask = buckets == i
+            if mask.any():
+                new_means[i] = samples[mask].mean(dim=0)
+
+        means = torch.where(zero_mask[:, None], means, new_means)
+
+    return means, bins
+
+
+class EuclideanCodebook(nn.Module):
+    """Codebook with Euclidean distance.
+    Args:
+        dim (int): Dimension.
+        codebook_size (int): Codebook size.
+        kmeans_init (bool): Whether to use k-means to initialize the codebooks.
+            If set to true, run the k-means algorithm on the first training batch and use
+            the learned centroids as initialization.
+        kmeans_iters (int): Number of iterations used for k-means algorithm at initialization.
+        decay (float): Decay for exponential moving average over the codebooks.
+        epsilon (float): Epsilon value for numerical stability.
+        threshold_ema_dead_code (int): Threshold for dead code expiration. Replace any codes
+            that have an exponential moving average cluster size less than the specified threshold with
+            randomly selected vector from the current batch.
+    """
+
+    def __init__(
+        self,
+        dim: int,
+        codebook_size: int,
+        kmeans_init: int = False,
+        kmeans_iters: int = 10,
+        decay: float = 0.99,
+        epsilon: float = 1e-5,
+        threshold_ema_dead_code: int = 2,
+    ):
+        super().__init__()
+        self.decay = decay
+        init_fn: tp.Union[tp.Callable[..., torch.Tensor], tp.Any] = uniform_init if not kmeans_init else torch.zeros
+        embed = init_fn(codebook_size, dim)
+
+        self.codebook_size = codebook_size
+
+        self.kmeans_iters = kmeans_iters
+        self.epsilon = epsilon
+        self.threshold_ema_dead_code = threshold_ema_dead_code
+
+        # Flag variable to indicate whether the codebook is initialized
+        self.register_buffer("inited", torch.Tensor([not kmeans_init]))
+        # Runing EMA cluster size/count: N_i^t in eq. (6) in vqvae paper
+        self.register_buffer("cluster_size", torch.zeros(codebook_size))
+        # Codebook
+        self.register_buffer("embed", embed)
+        # EMA codebook: eq. (7) in vqvae paper
+        self.register_buffer("embed_avg", embed.clone())
+
+    @torch.jit.ignore
+    def init_embed_(self, data):
+        """Initialize codebook.
+        Args:
+            data (tensor): [B * T, D].
+        """
+        if self.inited:
+            return
+
+        ## NOTE (snippet added by Songxiang Liu): gather data from all gpus
+        if dist.is_available() and dist.is_initialized():
+            # [B * T * world_size, D]
+            data = SyncFunction.apply(data)
+
+        embed, cluster_size = kmeans(data, self.codebook_size, self.kmeans_iters)
+        self.embed.data.copy_(embed)
+        self.embed_avg.data.copy_(embed.clone())
+        self.cluster_size.data.copy_(cluster_size)
+        self.inited.data.copy_(torch.Tensor([True]))
+        # Make sure all buffers across workers are in sync after initialization
+        broadcast_tensors(self.buffers())
+
+    def replace_(self, samples, mask):
+        modified_codebook = torch.where(mask[..., None], sample_vectors(samples, self.codebook_size), self.embed)
+        self.embed.data.copy_(modified_codebook)
+
+    def expire_codes_(self, batch_samples):
+        if self.threshold_ema_dead_code == 0:
+            return
+
+        expired_codes = self.cluster_size < self.threshold_ema_dead_code
+        if not torch.any(expired_codes):
+            return
+
+        ## NOTE (snippet added by Songxiang Liu): gather data from all gpus
+        if is_distributed():
+            # [B * T * world_size, D]
+            batch_samples = SyncFunction.apply(batch_samples)
+
+        batch_samples = rearrange(batch_samples, "... d -> (...) d")
+        self.replace_(batch_samples, mask=expired_codes)
+        broadcast_tensors(self.buffers())
+
+    def preprocess(self, x):
+        x = rearrange(x, "... d -> (...) d")
+        return x
+
+    def quantize(self, x):
+        embed = self.embed.t()
+        dist = -(x.pow(2).sum(1, keepdim=True) - 2 * x @ embed + embed.pow(2).sum(0, keepdim=True))
+        embed_ind = dist.max(dim=-1).indices
+        return embed_ind
+
+    def postprocess_emb(self, embed_ind, shape):
+        return embed_ind.view(*shape[:-1])
+
+    def dequantize(self, embed_ind):
+        quantize = F.embedding(embed_ind, self.embed)
+        return quantize
+
+    def encode(self, x):
+        shape = x.shape
+        # pre-process
+        x = self.preprocess(x)  # [B, T, D] -> [B*T, D]
+        # quantize
+        embed_ind = self.quantize(x)
+        # post-process
+        embed_ind = self.postprocess_emb(embed_ind, shape)
+        return embed_ind
+
+    def decode(self, embed_ind):
+        quantize = self.dequantize(embed_ind)
+        return quantize
+
+    def forward(self, x):
+        # shape: [B, T, D]
+        shape, dtype = x.shape, x.dtype
+        x = self.preprocess(x)  # [B, T, D] -> [B*T, D]
+
+        # Initialize codebook
+        self.init_embed_(x)
+
+        embed_ind = self.quantize(x)  # [B*T,]
+        embed_onehot = F.one_hot(embed_ind, self.codebook_size).type(dtype)  # [B*T, cb-size]
+        embed_ind = self.postprocess_emb(embed_ind, shape)  # [B, T]
+        quantize = self.dequantize(embed_ind)  # [B, T, D]
+
+        if self.training:
+            ### Update codebook by EMA
+            embed_onehot_sum = embed_onehot.sum(0)  # [cb-size,]
+            embed_sum = x.t() @ embed_onehot  # [D, cb-size]
+            if is_distributed():
+                dist.all_reduce(embed_onehot_sum)
+                dist.all_reduce(embed_sum)
+            # Update ema cluster count N_i^t, eq. (6) in vqvae paper
+            self.cluster_size.data.mul_(self.decay).add_(embed_onehot_sum, alpha=1 - self.decay)
+            # Update ema embed: eq. (7) in vqvae paper
+            self.embed_avg.data.mul_(self.decay).add_(embed_sum.t(), alpha=1 - self.decay)
+            # apply laplace smoothing
+            n = self.cluster_size.sum()
+            cluster_size = (self.cluster_size + self.epsilon) / (n + self.codebook_size * self.epsilon) * n
+            # Update ema embed: eq. (8) in vqvae paper
+            embed_normalized = self.embed_avg / cluster_size.unsqueeze(1)
+            self.embed.data.copy_(embed_normalized)
+
+            # We do the expiry of code at that point as buffers are in sync
+            # and all the workers will take the same decision.
+            self.expire_codes_(x)
+
+        return quantize, embed_ind
+
+
+class VectorQuantization(nn.Module):
+    """Vector quantization implementation.
+    Currently supports only euclidean distance.
+    Args:
+        dim (int): Dimension
+        codebook_size (int): Codebook size
+        codebook_dim (int): Codebook dimension. If not defined, uses the specified dimension in dim.
+        decay (float): Decay for exponential moving average over the codebooks.
+        epsilon (float): Epsilon value for numerical stability.
+        kmeans_init (bool): Whether to use kmeans to initialize the codebooks.
+        kmeans_iters (int): Number of iterations used for kmeans initialization.
+        threshold_ema_dead_code (int): Threshold for dead code expiration. Replace any codes
+            that have an exponential moving average cluster size less than the specified threshold with
+            randomly selected vector from the current batch.
+        commitment_weight (float): Weight for commitment loss.
+    """
+
+    def __init__(
+        self,
+        dim: int,
+        codebook_size: int,
+        codebook_dim: tp.Optional[int] = None,
+        decay: float = 0.99,
+        epsilon: float = 1e-5,
+        kmeans_init: bool = True,
+        kmeans_iters: int = 50,
+        threshold_ema_dead_code: int = 2,
+        commitment_weight: float = 1.0,
+    ):
+        super().__init__()
+        _codebook_dim: int = default(codebook_dim, dim)
+
+        requires_projection = _codebook_dim != dim
+        self.project_in = nn.Linear(dim, _codebook_dim) if requires_projection else nn.Identity()
+        self.project_out = nn.Linear(_codebook_dim, dim) if requires_projection else nn.Identity()
+
+        self.epsilon = epsilon
+        self.commitment_weight = commitment_weight
+
+        self._codebook = EuclideanCodebook(
+            dim=_codebook_dim,
+            codebook_size=codebook_size,
+            kmeans_init=kmeans_init,
+            kmeans_iters=kmeans_iters,
+            decay=decay,
+            epsilon=epsilon,
+            threshold_ema_dead_code=threshold_ema_dead_code,
+        )
+        self.codebook_size = codebook_size
+
+    @property
+    def codebook(self):
+        return self._codebook.embed
+
+    def encode(self, x):
+        x = rearrange(x, "b d n -> b n d")
+        x = self.project_in(x)
+        embed_in = self._codebook.encode(x)
+        return embed_in
+
+    def decode(self, embed_ind):
+        quantize = self._codebook.decode(embed_ind)
+        quantize = self.project_out(quantize)
+        quantize = rearrange(quantize, "b n d -> b d n")
+        return quantize
+
+    def forward(self, x):
+        device = x.device
+        x = x.transpose(1, 2).contiguous()  # [b d n] -> [b n d]
+        x = self.project_in(x)
+
+        quantize, embed_ind = self._codebook(x)
+
+        if self.training:
+            quantize = x + (quantize - x).detach()
+
+        loss = torch.tensor([0.0], device=device, requires_grad=self.training)
+
+        if self.training:
+            if self.commitment_weight > 0:
+                commit_loss = F.mse_loss(quantize.detach(), x)
+                loss = loss + commit_loss * self.commitment_weight
+
+        quantize = self.project_out(quantize)
+        quantize = quantize.transpose(1, 2).contiguous()  # [b n d] -> [b d n]
+        return quantize, embed_ind, loss
+
+
+class ResidualVectorQuantization(nn.Module):
+    """Residual vector quantization implementation.
+    Follows Algorithm 1. in https://arxiv.org/pdf/2107.03312.pdf
+    """
+
+    def __init__(self, *, num_quantizers, **kwargs):
+        super().__init__()
+        self.layers = nn.ModuleList([VectorQuantization(**kwargs) for _ in range(num_quantizers)])
+
+    def forward(self, x, n_q: tp.Optional[int] = None):
+        quantized_out = 0.0
+        residual = x
+
+        all_losses = []
+        all_indices = []
+
+        n_q = n_q or len(self.layers)
+
+        for layer in self.layers[:n_q]:
+            quantized, indices, loss = layer(residual)
+            residual = residual - quantized
+            quantized_out = quantized_out + quantized
+
+            all_indices.append(indices)
+            all_losses.append(loss)
+
+        out_losses, out_indices = map(torch.stack, (all_losses, all_indices))
+        return quantized_out, out_indices, out_losses
+
+    def encode(self, x: torch.Tensor, n_q: tp.Optional[int] = None) -> torch.Tensor:
+        residual = x
+        all_indices = []
+        n_q = n_q or len(self.layers)
+        for layer in self.layers[:n_q]:
+            indices = layer.encode(residual)
+            quantized = layer.decode(indices)
+            residual = residual - quantized
+            all_indices.append(indices)
+        out_indices = torch.stack(all_indices)
+        return out_indices
+
+    def decode(self, q_indices: torch.Tensor) -> torch.Tensor:
+        quantized_out = torch.tensor(0.0, device=q_indices.device)
+        for i, indices in enumerate(q_indices):
+            layer = self.layers[i]
+            quantized = layer.decode(indices)
+            quantized_out = quantized_out + quantized
+        return quantized_out

train-higgs-audio/boson_multimodal/audio_processing/quantization/ddp_utils.py

@@ -0,0 +1,197 @@
+import logging
+import random
+import subprocess
+from datetime import datetime
+
+import numpy as np
+import torch
+import torch.distributed as dist
+from torch.nn.parallel import DistributedDataParallel
+from torch.nn.parallel.distributed import _find_tensors
+import torch.optim
+import torch.utils.data
+from packaging import version
+from omegaconf import OmegaConf
+
+
+def set_random_seed(seed):
+    random.seed(seed)
+    np.random.seed(seed)
+    torch.manual_seed(seed)
+    torch.cuda.manual_seed_all(seed)
+
+
+def is_logging_process():
+    return not dist.is_initialized() or dist.get_rank() == 0
+
+
+def get_logger(cfg, name=None):
+    # log_file_path is used when unit testing
+    if is_logging_process():
+        logging.config.dictConfig(OmegaConf.to_container(cfg.job_logging_config, resolve=True))
+        return logging.getLogger(name)
+
+
+# from https://github.com/Lightning-AI/lightning-bolts/blob/5d61197cd2f491f69e238137a5edabe80ae14ad9/pl_bolts/models/self_supervised/simclr/simclr_module.py#L20
+class SyncFunction(torch.autograd.Function):
+    @staticmethod
+    # @torch.no_grad()
+    def forward(ctx, tensor):
+        ctx.batch_size = tensor.shape[0]
+
+        gathered_tensor = [torch.zeros_like(tensor) for _ in range(torch.distributed.get_world_size())]
+
+        torch.distributed.all_gather(gathered_tensor, tensor)
+        gathered_tensor = torch.cat(gathered_tensor, 0)
+
+        return gathered_tensor
+
+    @staticmethod
+    def backward(ctx, grad_output):
+        grad_input = grad_output.clone()
+        torch.distributed.all_reduce(grad_input, op=torch.distributed.ReduceOp.SUM, async_op=False)
+
+        idx_from = torch.distributed.get_rank() * ctx.batch_size
+        idx_to = (torch.distributed.get_rank() + 1) * ctx.batch_size
+        return grad_input[idx_from:idx_to]
+
+
+def get_timestamp():
+    return datetime.now().strftime("%y%m%d-%H%M%S")
+
+
+def get_commit_hash():
+    message = subprocess.check_output(["git", "rev-parse", "--short", "HEAD"])
+    return message.strip().decode("utf-8")
+
+
+class DDP(DistributedDataParallel):
+    """
+    Override the forward call in lightning so it goes to training and validation step respectively
+    """
+
+    def forward(self, *inputs, **kwargs):  # pragma: no cover
+        if version.parse(torch.__version__[:6]) < version.parse("1.11"):
+            self._sync_params()
+            inputs, kwargs = self.scatter(inputs, kwargs, self.device_ids)
+            assert len(self.device_ids) == 1
+            if self.module.training:
+                output = self.module.training_step(*inputs[0], **kwargs[0])
+            elif self.module.testing:
+                output = self.module.test_step(*inputs[0], **kwargs[0])
+            else:
+                output = self.module.validation_step(*inputs[0], **kwargs[0])
+            if torch.is_grad_enabled():
+                # We'll return the output object verbatim since it is a freeform
+                # object. We need to find any tensors in this object, though,
+                # because we need to figure out which parameters were used during
+                # this forward pass, to ensure we short circuit reduction for any
+                # unused parameters. Only if `find_unused_parameters` is set.
+                if self.find_unused_parameters:
+                    self.reducer.prepare_for_backward(list(_find_tensors(output)))
+                else:
+                    self.reducer.prepare_for_backward([])
+        else:
+            from torch.nn.parallel.distributed import (
+                logging,
+                Join,
+                _DDPSink,
+                _tree_flatten_with_rref,
+                _tree_unflatten_with_rref,
+            )
+
+            with torch.autograd.profiler.record_function("DistributedDataParallel.forward"):
+                if torch.is_grad_enabled() and self.require_backward_grad_sync:
+                    self.logger.set_runtime_stats_and_log()
+                    self.num_iterations += 1
+                    self.reducer.prepare_for_forward()
+
+                # Notify the join context that this process has not joined, if
+                # needed
+                work = Join.notify_join_context(self)
+                if work:
+                    self.reducer._set_forward_pass_work_handle(work, self._divide_by_initial_world_size)
+
+                # Calling _rebuild_buckets before forward compuation,
+                # It may allocate new buckets before deallocating old buckets
+                # inside _rebuild_buckets. To save peak memory usage,
+                # call _rebuild_buckets before the peak memory usage increases
+                # during forward computation.
+                # This should be called only once during whole training period.
+                if torch.is_grad_enabled() and self.reducer._rebuild_buckets():
+                    logging.info("Reducer buckets have been rebuilt in this iteration.")
+                    self._has_rebuilt_buckets = True
+
+                # sync params according to location (before/after forward) user
+                # specified as part of hook, if hook was specified.
+                buffer_hook_registered = hasattr(self, "buffer_hook")
+                if self._check_sync_bufs_pre_fwd():
+                    self._sync_buffers()
+
+                if self._join_config.enable:
+                    # Notify joined ranks whether they should sync in backwards pass or not.
+                    self._check_global_requires_backward_grad_sync(is_joined_rank=False)
+
+                inputs, kwargs = self.scatter(inputs, kwargs, self.device_ids)
+                if self.module.training:
+                    output = self.module.training_step(*inputs[0], **kwargs[0])
+                elif self.module.testing:
+                    output = self.module.test_step(*inputs[0], **kwargs[0])
+                else:
+                    output = self.module.validation_step(*inputs[0], **kwargs[0])
+
+                # sync params according to location (before/after forward) user
+                # specified as part of hook, if hook was specified.
+                if self._check_sync_bufs_post_fwd():
+                    self._sync_buffers()
+
+                if torch.is_grad_enabled() and self.require_backward_grad_sync:
+                    self.require_forward_param_sync = True
+                    # We'll return the output object verbatim since it is a freeform
+                    # object. We need to find any tensors in this object, though,
+                    # because we need to figure out which parameters were used during
+                    # this forward pass, to ensure we short circuit reduction for any
+                    # unused parameters. Only if `find_unused_parameters` is set.
+                    if self.find_unused_parameters and not self.static_graph:
+                        # Do not need to populate this for static graph.
+                        self.reducer.prepare_for_backward(list(_find_tensors(output)))
+                    else:
+                        self.reducer.prepare_for_backward([])
+                else:
+                    self.require_forward_param_sync = False
+
+            # TODO: DDPSink is currently enabled for unused parameter detection and
+            # static graph training for first iteration.
+            if (self.find_unused_parameters and not self.static_graph) or (
+                self.static_graph and self.num_iterations == 1
+            ):
+                state_dict = {
+                    "static_graph": self.static_graph,
+                    "num_iterations": self.num_iterations,
+                }
+
+                output_tensor_list, treespec, output_is_rref = _tree_flatten_with_rref(output)
+                output_placeholders = [None for _ in range(len(output_tensor_list))]
+                # Do not touch tensors that have no grad_fn, which can cause issues
+                # such as https://github.com/pytorch/pytorch/issues/60733
+                for i, output in enumerate(output_tensor_list):
+                    if torch.is_tensor(output) and output.grad_fn is None:
+                        output_placeholders[i] = output
+
+                # When find_unused_parameters=True, makes tensors which require grad
+                # run through the DDPSink backward pass. When not all outputs are
+                # used in loss, this makes those corresponding tensors receive
+                # undefined gradient which the reducer then handles to ensure
+                # param.grad field is not touched and we don't error out.
+                passthrough_tensor_list = _DDPSink.apply(
+                    self.reducer,
+                    state_dict,
+                    *output_tensor_list,
+                )
+                for i in range(len(output_placeholders)):
+                    if output_placeholders[i] is None:
+                        output_placeholders[i] = passthrough_tensor_list[i]
+
+                # Reconstruct output data structure.
+                output = _tree_unflatten_with_rref(output_placeholders, treespec, output_is_rref)
+        return output

train-higgs-audio/boson_multimodal/audio_processing/quantization/distrib.py

@@ -0,0 +1,123 @@
+# Copyright (c) Meta Platforms, Inc. and affiliates.
+# All rights reserved.
+#
+# This source code is licensed under the license found in the
+# LICENSE file in the root directory of this source tree.
+
+"""Torch distributed utilities."""
+
+import typing as tp
+
+import torch
+
+
+def rank():
+    if torch.distributed.is_initialized():
+        return torch.distributed.get_rank()
+    else:
+        return 0
+
+
+def world_size():
+    if torch.distributed.is_initialized():
+        return torch.distributed.get_world_size()
+    else:
+        return 1
+
+
+def is_distributed():
+    return world_size() > 1
+
+
+def all_reduce(tensor: torch.Tensor, op=torch.distributed.ReduceOp.SUM):
+    if is_distributed():
+        return torch.distributed.all_reduce(tensor, op)
+
+
+def _is_complex_or_float(tensor):
+    return torch.is_floating_point(tensor) or torch.is_complex(tensor)
+
+
+def _check_number_of_params(params: tp.List[torch.Tensor]):
+    # utility function to check that the number of params in all workers is the same,
+    # and thus avoid a deadlock with distributed all reduce.
+    if not is_distributed() or not params:
+        return
+    # print('params[0].device ', params[0].device)
+    tensor = torch.tensor([len(params)], device=params[0].device, dtype=torch.long)
+    all_reduce(tensor)
+    if tensor.item() != len(params) * world_size():
+        # If not all the workers have the same number, for at least one of them,
+        # this inequality will be verified.
+        raise RuntimeError(
+            f"Mismatch in number of params: ours is {len(params)}, at least one worker has a different one."
+        )
+
+
+def broadcast_tensors(tensors: tp.Iterable[torch.Tensor], src: int = 0):
+    """Broadcast the tensors from the given parameters to all workers.
+    This can be used to ensure that all workers have the same model to start with.
+    """
+    if not is_distributed():
+        return
+    tensors = [tensor for tensor in tensors if _is_complex_or_float(tensor)]
+    _check_number_of_params(tensors)
+    handles = []
+    for tensor in tensors:
+        handle = torch.distributed.broadcast(tensor.data, src=src, async_op=True)
+        handles.append(handle)
+    for handle in handles:
+        handle.wait()
+
+
+def sync_buffer(buffers, average=True):
+    """
+    Sync grad for buffers. If average is False, broadcast instead of averaging.
+    """
+    if not is_distributed():
+        return
+    handles = []
+    for buffer in buffers:
+        if torch.is_floating_point(buffer.data):
+            if average:
+                handle = torch.distributed.all_reduce(buffer.data, op=torch.distributed.ReduceOp.SUM, async_op=True)
+            else:
+                handle = torch.distributed.broadcast(buffer.data, src=0, async_op=True)
+            handles.append((buffer, handle))
+    for buffer, handle in handles:
+        handle.wait()
+        if average:
+            buffer.data /= world_size
+
+
+def sync_grad(params):
+    """
+    Simpler alternative to DistributedDataParallel, that doesn't rely
+    on any black magic. For simple models it can also be as fast.
+    Just call this on your model parameters after the call to backward!
+    """
+    if not is_distributed():
+        return
+    handles = []
+    for p in params:
+        if p.grad is not None:
+            handle = torch.distributed.all_reduce(p.grad.data, op=torch.distributed.ReduceOp.SUM, async_op=True)
+            handles.append((p, handle))
+    for p, handle in handles:
+        handle.wait()
+        p.grad.data /= world_size()
+
+
+def average_metrics(metrics: tp.Dict[str, float], count=1.0):
+    """Average a dictionary of metrics across all workers, using the optional
+    `count` as unormalized weight.
+    """
+    if not is_distributed():
+        return metrics
+    keys, values = zip(*metrics.items())
+    device = "cuda" if torch.cuda.is_available() else "cpu"
+    tensor = torch.tensor(list(values) + [1], device=device, dtype=torch.float32)
+    tensor *= count
+    all_reduce(tensor)
+    averaged = (tensor[:-1] / tensor[-1]).cpu().tolist()
+    return dict(zip(keys, averaged))

train-higgs-audio/boson_multimodal/audio_processing/quantization/vq.py

@@ -0,0 +1,116 @@
+# Copyright (c) Meta Platforms, Inc. and affiliates.
+# All rights reserved.
+#
+# This source code is licensed under the license found in the
+# LICENSE file in the root directory of this source tree.
+
+"""Residual vector quantizer implementation."""
+
+from dataclasses import dataclass, field
+import math
+import typing as tp
+
+import torch
+from torch import nn
+
+# from .core_vq import ResidualVectorQuantization
+from .core_vq_lsx_version import ResidualVectorQuantization
+
+
+@dataclass
+class QuantizedResult:
+    quantized: torch.Tensor
+    codes: torch.Tensor
+    bandwidth: torch.Tensor  # bandwidth in kb/s used, per batch item.
+    penalty: tp.Optional[torch.Tensor] = None
+    metrics: dict = field(default_factory=dict)
+
+
+class ResidualVectorQuantizer(nn.Module):
+    """Residual Vector Quantizer.
+    Args:
+        dimension (int): Dimension of the codebooks.
+        n_q (int): Number of residual vector quantizers used.
+        bins (int): Codebook size.
+        decay (float): Decay for exponential moving average over the codebooks.
+        kmeans_init (bool): Whether to use kmeans to initialize the codebooks.
+        kmeans_iters (int): Number of iterations used for kmeans initialization.
+        threshold_ema_dead_code (int): Threshold for dead code expiration. Replace any codes
+            that have an exponential moving average cluster size less than the specified threshold with
+            randomly selected vector from the current batch.
+    """
+
+    def __init__(
+        self,
+        dimension: int = 256,
+        codebook_dim: int = None,
+        n_q: int = 8,
+        bins: int = 1024,
+        decay: float = 0.99,
+        kmeans_init: bool = True,
+        kmeans_iters: int = 50,
+        threshold_ema_dead_code: int = 2,
+    ):
+        super().__init__()
+        self.n_q = n_q
+        self.dimension = dimension
+        self.codebook_dim = codebook_dim
+        self.bins = bins
+        self.decay = decay
+        self.kmeans_init = kmeans_init
+        self.kmeans_iters = kmeans_iters
+        self.threshold_ema_dead_code = threshold_ema_dead_code
+        self.vq = ResidualVectorQuantization(
+            dim=self.dimension,
+            codebook_dim=self.codebook_dim,
+            codebook_size=self.bins,
+            num_quantizers=self.n_q,
+            decay=self.decay,
+            kmeans_init=self.kmeans_init,
+            kmeans_iters=self.kmeans_iters,
+            threshold_ema_dead_code=self.threshold_ema_dead_code,
+        )
+
+    def forward(self, x: torch.Tensor, sample_rate: int, bandwidth: tp.Optional[float] = None):  # -> QuantizedResult:
+        """Residual vector quantization on the given input tensor.
+        Args:
+            x (torch.Tensor): Input tensor.
+            sample_rate (int): Sample rate of the input tensor.
+            bandwidth (float): Target bandwidth.
+        Returns:
+            QuantizedResult:
+                The quantized (or approximately quantized) representation with
+                the associated bandwidth and any penalty term for the loss.
+        """
+        bw_per_q = self.get_bandwidth_per_quantizer(sample_rate)
+        n_q = self.get_num_quantizers_for_bandwidth(sample_rate, bandwidth)
+        quantized, codes, commit_loss = self.vq(x, n_q=n_q)
+        bw = torch.tensor(n_q * bw_per_q).to(x)
+        return quantized, codes, bw, torch.mean(commit_loss)
+        # return QuantizedResult(quantized, codes, bw, penalty=torch.mean(commit_loss))
+
+    def get_num_quantizers_for_bandwidth(self, sample_rate: int, bandwidth: tp.Optional[float] = None) -> int:
+        """Return n_q based on specified target bandwidth."""
+        bw_per_q = self.get_bandwidth_per_quantizer(sample_rate)
+        n_q = self.n_q
+        if bandwidth and bandwidth > 0.0:
+            n_q = int(max(1, math.floor(bandwidth / bw_per_q)))
+        return n_q
+
+    def get_bandwidth_per_quantizer(self, sample_rate: int):
+        """Return bandwidth per quantizer for a given input sample rate."""
+        return math.log2(self.bins) * sample_rate / 1000
+
+    def encode(self, x: torch.Tensor, sample_rate: int, bandwidth: tp.Optional[float] = None) -> torch.Tensor:
+        """Encode a given input tensor with the specified sample rate at the given bandwidth.
+        The RVQ encode method sets the appropriate number of quantizer to use
+        and returns indices for each quantizer.
+        """
+        n_q = self.get_num_quantizers_for_bandwidth(sample_rate, bandwidth)
+        codes = self.vq.encode(x, n_q=n_q)
+        return codes
+
+    def decode(self, codes: torch.Tensor) -> torch.Tensor:
+        """Decode the given codes to the quantized representation."""
+        quantized = self.vq.decode(codes)
+        return quantized

train-higgs-audio/boson_multimodal/audio_processing/semantic_module.py

@@ -0,0 +1,282 @@
+# Based on code from: https://github.com/zhenye234/xcodec
+# Licensed under MIT License
+# Modifications by BosonAI
+
+import torch
+import torch.nn as nn
+
+
+class Conv1d1x1(nn.Conv1d):
+    """1x1 Conv1d."""
+
+    def __init__(self, in_channels, out_channels, bias=True):
+        super(Conv1d1x1, self).__init__(in_channels, out_channels, kernel_size=1, bias=bias)
+
+
+class Conv1d(nn.Module):
+    def __init__(
+        self,
+        in_channels: int,
+        out_channels: int,
+        kernel_size: int,
+        stride: int = 1,
+        padding: int = -1,
+        dilation: int = 1,
+        groups: int = 1,
+        bias: bool = True,
+    ):
+        super().__init__()
+        self.in_channels = in_channels
+        self.out_channels = out_channels
+        self.kernel_size = kernel_size
+        if padding < 0:
+            padding = (kernel_size - 1) // 2 * dilation
+        self.dilation = dilation
+        self.conv = nn.Conv1d(
+            in_channels=in_channels,
+            out_channels=out_channels,
+            kernel_size=kernel_size,
+            stride=stride,
+            padding=padding,
+            dilation=dilation,
+            groups=groups,
+            bias=bias,
+        )
+
+    def forward(self, x):
+        """
+        Args:
+            x (Tensor): Float tensor variable with the shape  (B, C, T).
+        Returns:
+            Tensor: Float tensor variable with the shape (B, C, T).
+        """
+        x = self.conv(x)
+        return x
+
+
+class ResidualUnit(nn.Module):
+    def __init__(
+        self,
+        in_channels: int,
+        out_channels: int,
+        kernel_size=3,
+        dilation=1,
+        bias=False,
+        nonlinear_activation="ELU",
+        nonlinear_activation_params={},
+    ):
+        super().__init__()
+        self.activation = getattr(nn, nonlinear_activation)(**nonlinear_activation_params)
+        self.conv1 = Conv1d(
+            in_channels=in_channels,
+            out_channels=out_channels,
+            kernel_size=kernel_size,
+            stride=1,
+            dilation=dilation,
+            bias=bias,
+        )
+        self.conv2 = Conv1d1x1(out_channels, out_channels, bias)
+
+    def forward(self, x):
+        y = self.conv1(self.activation(x))
+        y = self.conv2(self.activation(y))
+        return x + y
+
+
+class ConvTranspose1d(nn.Module):
+    def __init__(
+        self,
+        in_channels: int,
+        out_channels: int,
+        kernel_size: int,
+        stride: int,
+        padding=-1,
+        output_padding=-1,
+        groups=1,
+        bias=True,
+    ):
+        super().__init__()
+        if padding < 0:
+            padding = (stride + 1) // 2
+        if output_padding < 0:
+            output_padding = 1 if stride % 2 else 0
+        self.deconv = nn.ConvTranspose1d(
+            in_channels=in_channels,
+            out_channels=out_channels,
+            kernel_size=kernel_size,
+            stride=stride,
+            padding=padding,
+            output_padding=output_padding,
+            groups=groups,
+            bias=bias,
+        )
+
+    def forward(self, x):
+        """
+        Args:
+            x (Tensor): Float tensor variable with the shape  (B, C, T).
+        Returns:
+            Tensor: Float tensor variable with the shape (B, C', T').
+        """
+        x = self.deconv(x)
+        return x
+
+
+class EncoderBlock(nn.Module):
+    def __init__(
+        self, in_channels: int, out_channels: int, stride: int, dilations=(1, 1), unit_kernel_size=3, bias=True
+    ):
+        super().__init__()
+        self.res_units = torch.nn.ModuleList()
+        for dilation in dilations:
+            self.res_units += [ResidualUnit(in_channels, in_channels, kernel_size=unit_kernel_size, dilation=dilation)]
+        self.num_res = len(self.res_units)
+
+        self.conv = Conv1d(
+            in_channels=in_channels,
+            out_channels=out_channels,
+            kernel_size=3 if stride == 1 else (2 * stride),  # special case: stride=1, do not use kernel=2
+            stride=stride,
+            bias=bias,
+        )
+
+    def forward(self, x):
+        for idx in range(self.num_res):
+            x = self.res_units[idx](x)
+        x = self.conv(x)
+        return x
+
+
+class Encoder(nn.Module):
+    def __init__(
+        self,
+        input_channels: int,
+        encode_channels: int,
+        channel_ratios=(1, 1),
+        strides=(1, 1),
+        kernel_size=3,
+        bias=True,
+        block_dilations=(1, 1),
+        unit_kernel_size=3,
+    ):
+        super().__init__()
+        assert len(channel_ratios) == len(strides)
+
+        self.conv = Conv1d(
+            in_channels=input_channels, out_channels=encode_channels, kernel_size=kernel_size, stride=1, bias=False
+        )
+        self.conv_blocks = torch.nn.ModuleList()
+        in_channels = encode_channels
+        for idx, stride in enumerate(strides):
+            out_channels = int(encode_channels * channel_ratios[idx])  # could be float
+            self.conv_blocks += [
+                EncoderBlock(
+                    in_channels,
+                    out_channels,
+                    stride,
+                    dilations=block_dilations,
+                    unit_kernel_size=unit_kernel_size,
+                    bias=bias,
+                )
+            ]
+            in_channels = out_channels
+        self.num_blocks = len(self.conv_blocks)
+        self.out_channels = out_channels
+
+    def forward(self, x):
+        x = self.conv(x)
+        for i in range(self.num_blocks):
+            x = self.conv_blocks[i](x)
+        return x
+
+
+class DecoderBlock(nn.Module):
+    """Decoder block (no up-sampling)"""
+
+    def __init__(
+        self, in_channels: int, out_channels: int, stride: int, dilations=(1, 1), unit_kernel_size=3, bias=True
+    ):
+        super().__init__()
+
+        if stride == 1:
+            self.conv = Conv1d(
+                in_channels=in_channels,
+                out_channels=out_channels,
+                kernel_size=3,  # fix kernel=3 when stride=1 for unchanged shape
+                stride=stride,
+                bias=bias,
+            )
+        else:
+            self.conv = ConvTranspose1d(
+                in_channels=in_channels,
+                out_channels=out_channels,
+                kernel_size=(2 * stride),
+                stride=stride,
+                bias=bias,
+            )
+
+        self.res_units = torch.nn.ModuleList()
+        for idx, dilation in enumerate(dilations):
+            self.res_units += [
+                ResidualUnit(out_channels, out_channels, kernel_size=unit_kernel_size, dilation=dilation)
+            ]
+        self.num_res = len(self.res_units)
+
+    def forward(self, x):
+        x = self.conv(x)
+        for idx in range(self.num_res):
+            x = self.res_units[idx](x)
+        return x
+
+
+class Decoder(nn.Module):
+    def __init__(
+        self,
+        code_dim: int,
+        output_channels: int,
+        decode_channels: int,
+        channel_ratios=(1, 1),
+        strides=(1, 1),
+        kernel_size=3,
+        bias=True,
+        block_dilations=(1, 1),
+        unit_kernel_size=3,
+    ):
+        super().__init__()
+        assert len(channel_ratios) == len(strides)
+
+        self.conv1 = Conv1d(
+            in_channels=code_dim,
+            out_channels=int(decode_channels * channel_ratios[0]),
+            kernel_size=kernel_size,
+            stride=1,
+            bias=False,
+        )
+
+        self.conv_blocks = torch.nn.ModuleList()
+        for idx, stride in enumerate(strides):
+            in_channels = int(decode_channels * channel_ratios[idx])
+            if idx < (len(channel_ratios) - 1):
+                out_channels = int(decode_channels * channel_ratios[idx + 1])
+            else:
+                out_channels = decode_channels
+            self.conv_blocks += [
+                DecoderBlock(
+                    in_channels,
+                    out_channels,
+                    stride,
+                    dilations=block_dilations,
+                    unit_kernel_size=unit_kernel_size,
+                    bias=bias,
+                )
+            ]
+        self.num_blocks = len(self.conv_blocks)
+
+        self.conv2 = Conv1d(out_channels, output_channels, kernel_size, 1, bias=False)
+
+    def forward(self, z):
+        x = self.conv1(z)
+        for i in range(self.num_blocks):
+            x = self.conv_blocks[i](x)
+        x = self.conv2(x)
+        return x

train-higgs-audio/boson_multimodal/constants.py

@@ -0,0 +1,3 @@
+AUDIO_IN_TOKEN = "<|AUDIO|>"
+AUDIO_OUT_TOKEN = "<|AUDIO_OUT|>"
+EOS_TOKEN = "<|end_of_text|>"

train-higgs-audio/boson_multimodal/data_collator/higgs_audio_collator.py

@@ -0,0 +1,509 @@
+import librosa
+import torch
+import torch.nn.functional as F
+import math
+from typing import List, Tuple
+
+from dataclasses import dataclass
+from typing import List, Optional
+from transformers.models.whisper.processing_whisper import WhisperProcessor
+
+from ..dataset.chatml_dataset import ChatMLDatasetSample
+from ..model.higgs_audio.utils import build_delay_pattern_mask
+
+
+def _ceil_to_nearest(n, round_to):
+    return (n + round_to - 1) // round_to * round_to
+
+
+def _ceil_to_next_power_of_two(self, x):
+    return 1 if x == 0 else 2 ** (x - 1).bit_length()
+
+
+@dataclass
+class HiggsAudioBatchInput:
+    input_ids: torch.LongTensor  # shape (bsz, seq_len).
+    attention_mask: torch.Tensor  # shape (bsz, seq_len).
+    audio_features: Optional[torch.Tensor]  # shape (num_audio_in, feature_dim, max_mel_seq_len).
+    audio_feature_attention_mask: Optional[torch.Tensor]  # shape (num_audio_in, max_mel_seq_len).
+    audio_out_ids: Optional[torch.LongTensor]  # shape (num_codebooks, audio_out_total_length)
+    audio_out_ids_start: Optional[torch.LongTensor]  # shape (num_audio_out,)
+    # The audio_out_ids_start_group_loc has the same length as audio_out_ids_start. It is used to recover group location in a batch for an audio segment
+    # Currently, we concatenante audio segments along dim 0 to handle variadic audio segment length. However, in the alignment stage, we need the location information
+    # For example,
+    #  audio_out_ids_start = [0, 2, 4, 8]; and the first two audio segments come from the same sample in a batch, and other two come from different samples.
+    #  This is a batch of 3 samples, then we will have the group location as:
+    #  audio_out_ids_start_group_loc = [0, 0, 1, 2]
+    audio_out_ids_start_group_loc: Optional[
+        torch.LongTensor
+    ]  # shape (num_audio_out,), specify which a sample's group location in the batch
+    audio_in_ids: Optional[torch.LongTensor]  # shape (num_codebooks, audio_in_total_length)
+    audio_in_ids_start: Optional[torch.LongTensor]  # shape (num_audio_in,)
+    label_ids: Optional[torch.LongTensor]  # shape (bsz, seq_len)
+    label_audio_ids: Optional[torch.LongTensor]  # shape (num_codebooks, audio_out_total_length)
+    reward: Optional[float] = None
+
+
+class HiggsAudioSampleCollator:
+    """Sample collator for Higgs-Audio model.
+
+    Args:
+        whisper_processor (WhisperProcessor): The whisper processor.
+        audio_in_token_id (int): The token id for audio-in.
+        audio_out_token_id (int): The token id for audio-out.
+        pad_token_id (int): The token id for padding.
+        audio_stream_bos_id (int): The token id for audio-stream beginning of sentence.
+        audio_stream_eos_id (int): The token id for audio-stream end of sentence.
+        round_to (int): The round-to value.
+        pad_left (bool): Whether to pad left.
+        return_audio_in_tokens (bool): Whether to return audio-in tokens.
+        use_delay_pattern (bool): Whether to use delay pattern.
+        disable_audio_codes_transform (bool): Whether to add bos and eos tokens to audio codes.
+        chunk_size_seconds (int): The chunk size in seconds.
+        add_new_bos_eos_for_long_chunk (bool): Whether to add new bos and eos tokens for long chunks.
+        mask_audio_out_token_label (bool): Whether to always mask the label associated with <|AUDIO_OUT|> token. Since we will always have `<|AUDIO_OUT|>` after `<|audio_bos|>`, we can safely mask <|AUDIO_OUT|>.
+
+    """
+
+    def __init__(
+        self,
+        whisper_processor: WhisperProcessor,
+        audio_in_token_id,
+        audio_out_token_id,
+        pad_token_id,
+        audio_stream_bos_id,
+        audio_stream_eos_id,
+        round_to=8,
+        pad_left=False,
+        encode_whisper_embed=True,
+        return_audio_in_tokens=True,
+        audio_num_codebooks=None,
+        use_delay_pattern=False,
+        disable_audio_codes_transform=False,
+        chunk_size_seconds=30,  # Maximum duration for each chunk
+        add_new_bos_eos_for_long_chunk=True,
+        mask_audio_out_token_label=True,
+    ):
+        self.whisper_processor = whisper_processor
+        self.round_to = round_to
+        self.pad_left = pad_left
+        self.audio_in_token_id = audio_in_token_id
+        self.audio_out_token_id = audio_out_token_id
+        self.audio_stream_bos_id = audio_stream_bos_id
+        self.audio_stream_eos_id = audio_stream_eos_id
+        self.pad_token_id = pad_token_id
+        self.encode_whisper_embed = encode_whisper_embed
+        self.return_audio_in_tokens = return_audio_in_tokens
+        self.audio_num_codebooks = audio_num_codebooks
+        self.use_delay_pattern = use_delay_pattern
+        if encode_whisper_embed:
+            self.chunk_size_seconds = chunk_size_seconds
+            self.chunk_size_samples = int(chunk_size_seconds * whisper_processor.feature_extractor.sampling_rate)
+        else:
+            self.chunk_size_seconds = None
+            self.chunk_size_samples = None
+        self.disable_audio_codes_transform = disable_audio_codes_transform
+        self.add_new_bos_eos_for_long_chunk = add_new_bos_eos_for_long_chunk
+        self.mask_audio_out_token_label = mask_audio_out_token_label
+
+    def _process_and_duplicate_audio_tokens(
+        self, input_ids: torch.Tensor, audio_idx: int, wv: torch.Tensor, sr: int, labels: Optional[torch.Tensor] = None
+    ) -> Tuple[torch.Tensor, torch.Tensor, int]:
+        """Process long audio and duplicate corresponding audio tokens.
+
+        Args:
+            input_ids: Input token ids
+            audio_idx: Index of the audio token in the sequence
+            wv: Audio waveform
+            sr: Sample rate
+            labels: Optional label ids to be duplicated alongside input ids
+
+        Returns:
+            Tuple of:
+                - New input ids with duplicated audio tokens
+                - New label ids (if labels were provided) or None
+                - Number of chunks created
+        """
+        # Calculate number of chunks needed
+        total_samples = len(wv)
+        num_chunks = math.ceil(total_samples / self.chunk_size_samples)
+
+        if num_chunks <= 1:
+            return input_ids, labels, 1
+
+        # Get the three tokens: <|audio_bos|><|AUDIO|><|audio_eos|>
+        audio_token_seq = input_ids[audio_idx - 1 : audio_idx + 2]
+        # Duplicate sequence for each chunk
+        duplicated_sequence = audio_token_seq.repeat(num_chunks)
+
+        # Create new input_ids with duplicated tokens
+        new_input_ids = torch.cat([input_ids[: audio_idx - 1], duplicated_sequence, input_ids[audio_idx + 2 :]])
+
+        # If labels are provided, duplicate them as well
+        new_labels = None
+        if labels is not None:
+            label_seq = labels[audio_idx - 1 : audio_idx + 2]
+            duplicated_labels = label_seq.repeat(num_chunks)
+            new_labels = torch.cat([labels[: audio_idx - 1], duplicated_labels, labels[audio_idx + 2 :]])
+
+        return new_input_ids, new_labels, num_chunks
+
+    def __call__(self, batch: List[ChatMLDatasetSample]):
+        """Collate the input data with support for long audio processing."""
+
+        label_ids = None
+        label_audio_ids = None
+        if all([ele.label_ids is None for ele in batch]):
+            return_labels = False
+        else:
+            return_labels = True
+
+        if self.encode_whisper_embed:
+            # Process each sample in the batch to handle long audio
+            # TODO(?) The implementation here can be optimized.
+            processed_batch = []
+            for i in range(len(batch)):
+                sample = batch[i]
+                audio_in_mask = sample.input_ids == self.audio_in_token_id
+                audio_in_indices = torch.where(audio_in_mask)[0]
+                audio_out_mask = sample.input_ids == self.audio_out_token_id
+
+                # Process each audio token and duplicate if needed
+                modified_input_ids = sample.input_ids
+                modified_labels = sample.label_ids if return_labels else None
+                modified_waveforms_concat = []
+                modified_waveforms_start = []
+                modified_sample_rate = []
+                offset = 0  # Track position changes from duplicating tokens
+                curr_wv_offset = 0
+
+                # Process input audio tokens
+                for idx, audio_idx in enumerate(audio_in_indices):
+                    # Get the audio for this token
+                    wv, sr = sample.get_wv(idx)  # Use idx since we want the original audio index
+                    if sr != self.whisper_processor.feature_extractor.sampling_rate:
+                        resampled_wv = librosa.resample(
+                            wv.cpu().numpy(),
+                            orig_sr=sr,
+                            target_sr=self.whisper_processor.feature_extractor.sampling_rate,
+                        )
+                    else:
+                        resampled_wv = wv.cpu().numpy()
+                    wv = torch.tensor(resampled_wv, device=wv.device)
+                    sr = self.whisper_processor.feature_extractor.sampling_rate
+
+                    # Process and duplicate tokens if necessary
+                    token_pos = audio_idx + offset
+                    modified_input_ids, modified_labels, num_chunks = self._process_and_duplicate_audio_tokens(
+                        modified_input_ids, token_pos, wv, sr, modified_labels
+                    )
+
+                    # Update audio data
+                    for chunk_idx in range(num_chunks):
+                        chunk_start = chunk_idx * self.chunk_size_samples
+                        chunk_end = min((chunk_idx + 1) * self.chunk_size_samples, len(wv))
+                        chunk_wv = wv[chunk_start:chunk_end]
+                        modified_waveforms_concat.append(chunk_wv)
+                        modified_waveforms_start.append(curr_wv_offset)
+                        curr_wv_offset += len(chunk_wv)
+                        modified_sample_rate.append(sr)
+
+                    # Update offset for next iteration
+                    offset += (num_chunks - 1) * 3  # Each new chunk adds 3 more tokens
+
+                # Create new sample with modified tokens and audio data
+                processed_sample = ChatMLDatasetSample(
+                    input_ids=modified_input_ids,
+                    label_ids=modified_labels if return_labels else sample.label_ids,
+                    audio_ids_concat=sample.audio_ids_concat,
+                    audio_ids_start=sample.audio_ids_start,
+                    audio_waveforms_concat=torch.cat(modified_waveforms_concat)
+                    if modified_waveforms_concat
+                    else sample.audio_waveforms_concat,
+                    audio_waveforms_start=torch.tensor(modified_waveforms_start, dtype=torch.long)
+                    if modified_waveforms_start
+                    else sample.audio_waveforms_start,
+                    audio_sample_rate=torch.tensor(modified_sample_rate)
+                    if modified_sample_rate
+                    else sample.audio_sample_rate,
+                    audio_speaker_indices=torch.tensor([]),
+                    # FIXME(sxjscience): The logic here is not correct for audio_label_ids_concat.
+                    audio_label_ids_concat=sample.audio_label_ids_concat,
+                )
+                # audio_in_chunk_len = len(torch.where(modified_input_ids == self.audio_in_token_id)[0])
+                # assert audio_in_chunk_len == processed_sample.num_audios(), f"Mismatch: audio_in_chunk_len={audio_in_chunk_len}, processed_sample.num_audios()={processed_sample.num_audios()}"
+                processed_batch.append(processed_sample)
+        else:
+            processed_batch = batch
+
+        # Get the max sequence length based on processed batch
+        max_seq_length = _ceil_to_nearest(max([len(sample.input_ids) for sample in processed_batch]), self.round_to)
+
+        # Get the ids for audio-in and audio-out for each batch
+        audio_in_wv_l = []
+        audio_in_ids_l = []
+        audio_out_ids_l = []
+        audio_out_ids_group_loc_l = []
+        audio_in_label_ids_l = None
+        audio_out_label_ids_l = None
+        reward_l = []
+
+        if return_labels:
+            audio_out_no_train_flag = []  # Whether the audio-out data should be trained on or not.
+
+        # Process the audio inputs and outputs
+        for i in range(len(processed_batch)):
+            audio_in_mask = processed_batch[i].input_ids == self.audio_in_token_id
+            audio_out_mask = processed_batch[i].input_ids == self.audio_out_token_id
+            audio_ids = torch.ones_like(processed_batch[i].input_ids)
+            audio_ids[audio_in_mask ^ audio_out_mask] = torch.cumsum(audio_ids[audio_in_mask ^ audio_out_mask], 0) - 1
+            audio_in_ids = audio_ids[audio_in_mask]
+            audio_out_ids = audio_ids[audio_out_mask]
+
+            if return_labels:
+                audio_out_no_train_flag.append(processed_batch[i].label_ids[audio_out_mask] < 0)
+                if self.mask_audio_out_token_label:
+                    processed_batch[i].label_ids[audio_out_mask] = -100
+
+            # Process audio inputs
+            if self.return_audio_in_tokens:
+                audio_in_ids_l.extend(
+                    [processed_batch[i].get_audio_codes(idx)[: self.audio_num_codebooks, :] for idx in audio_in_ids]
+                )
+                if processed_batch[i].audio_label_ids_concat is not None:
+                    if audio_in_label_ids_l is None:
+                        audio_in_label_ids_l = []
+                    audio_in_label_ids_l.extend(
+                        [
+                            processed_batch[i].get_audio_codes_labels(idx)[: self.audio_num_codebooks, :]
+                            for idx in audio_in_ids
+                        ]
+                    )
+
+            audio_out_ids_l.extend(
+                [processed_batch[i].get_audio_codes(idx)[: self.audio_num_codebooks, :] for idx in audio_out_ids]
+            )
+            audio_out_ids_group_loc_l.append(i)
+            if processed_batch[i].reward is not None:
+                reward_l.append(processed_batch[i].reward)
+
+            if processed_batch[i].audio_label_ids_concat is not None:
+                if audio_out_label_ids_l is None:
+                    audio_out_label_ids_l = []
+                audio_out_label_ids_l.extend(
+                    [
+                        processed_batch[i].get_audio_codes_labels(idx)[: self.audio_num_codebooks, :]
+                        for idx in audio_out_ids
+                    ]
+                )
+
+            if self.encode_whisper_embed:
+                for idx in audio_in_ids:
+                    wv, sr = processed_batch[i].get_wv(idx)
+                    resampled_wv = wv.cpu().numpy()
+                    # Split long audio into chunks
+                    total_samples = len(resampled_wv)
+                    for chunk_start in range(0, total_samples, self.chunk_size_samples):
+                        chunk_end = min(chunk_start + self.chunk_size_samples, total_samples)
+                        chunk = resampled_wv[chunk_start:chunk_end]
+                        audio_in_wv_l.append(chunk)
+            # assert len(audio_in_wv_l) == processed_batch[i].num_audios(), \
+            #     f"Assertion failed: Mismatch in number of audios. " \
+            #     f"Expected {processed_batch[i].num_audios()}, but got {len(audio_in_wv_l)} at index {i}."
+
+        if return_labels:
+            audio_out_no_train_flag = torch.cat(audio_out_no_train_flag, dim=0)
+
+        # Process all audio features
+        if len(audio_in_wv_l) > 0:
+            feature_ret = self.whisper_processor.feature_extractor(
+                audio_in_wv_l,
+                sampling_rate=self.whisper_processor.feature_extractor.sampling_rate,
+                return_attention_mask=True,
+                padding="max_length",
+            )
+            audio_features = torch.from_numpy(feature_ret["input_features"])
+            audio_feature_attention_mask = torch.from_numpy(feature_ret["attention_mask"])
+        else:
+            if self.encode_whisper_embed:
+                audio_features = torch.zeros(
+                    (
+                        0,
+                        self.whisper_processor.feature_extractor.feature_size,
+                        self.whisper_processor.feature_extractor.nb_max_frames,
+                    ),
+                    dtype=torch.float32,
+                )
+                audio_feature_attention_mask = torch.zeros(
+                    (0, self.whisper_processor.feature_extractor.nb_max_frames), dtype=torch.int32
+                )
+            else:
+                audio_features = None
+                audio_feature_attention_mask = None
+
+        # Process audio input tokens
+        if len(audio_in_ids_l) > 0:
+            # Append audio-stream-bos and eos tokens
+            new_audio_in_ids_l = []
+            for ele in audio_in_ids_l:
+                if self.disable_audio_codes_transform:
+                    # Do not add audio-stream-bos or eos tokens.
+                    # This may indicate that the sample comes from ConstantLengthDatasetWithBuffer.
+                    audio_codes = ele
+                else:
+                    audio_codes = torch.cat(
+                        [
+                            torch.full((ele.shape[0], 1), self.audio_stream_bos_id, dtype=torch.long),
+                            ele,
+                            torch.full((ele.shape[0], 1), self.audio_stream_eos_id, dtype=torch.long),
+                        ],
+                        dim=1,
+                    )
+                    if self.use_delay_pattern:
+                        audio_codes = build_delay_pattern_mask(
+                            audio_codes.unsqueeze(0),
+                            bos_token_id=self.audio_stream_bos_id,
+                            pad_token_id=self.audio_stream_eos_id,
+                        )[0].squeeze(0)
+                new_audio_in_ids_l.append(audio_codes)
+            audio_in_ids = torch.cat(new_audio_in_ids_l, dim=1).long()
+            audio_in_ids_start = torch.cumsum(
+                torch.tensor([0] + [audio_codes.shape[1] for audio_codes in new_audio_in_ids_l[:-1]]), dim=0
+            )
+        else:
+            audio_in_ids = torch.zeros((0, 0), dtype=torch.long)
+            audio_in_ids_start = torch.zeros(0, dtype=torch.long)
+
+        # Process audio output tokens
+        audio_out_ids_start_group_loc = None
+        if len(audio_out_ids_l) > 0:
+            new_audio_out_ids_l = []
+            label_audio_ids_l = []
+            for idx, ele in enumerate(audio_out_ids_l):
+                if self.disable_audio_codes_transform:
+                    # Do not add audio-stream-bos or eos tokens.
+                    # This may indicate that the sample comes from ConstantLengthDatasetWithBuffer.
+                    audio_codes = ele
+                    if return_labels:
+                        label_audio_ids = audio_out_label_ids_l[idx]
+                else:
+                    audio_codes = torch.cat(
+                        [
+                            torch.full((ele.shape[0], 1), self.audio_stream_bos_id, dtype=torch.long).to(ele.device),
+                            ele,
+                            torch.full((ele.shape[0], 1), self.audio_stream_eos_id, dtype=torch.long).to(ele.device),
+                        ],
+                        dim=1,
+                    )
+                    if return_labels:
+                        label_audio_ids = torch.cat(
+                            [
+                                torch.full((ele.shape[0], 1), -100, dtype=torch.long).to(ele.device),
+                                ele,
+                                torch.full((ele.shape[0], 1), self.audio_stream_eos_id, dtype=torch.long).to(ele.device),
+                            ],
+                            dim=1,
+                        )
+                    if self.use_delay_pattern:
+                        audio_codes = build_delay_pattern_mask(
+                            audio_codes.unsqueeze(0),
+                            bos_token_id=self.audio_stream_bos_id,
+                            pad_token_id=self.audio_stream_eos_id,
+                        )[0].squeeze(0)
+                        if return_labels:
+                            label_audio_ids = build_delay_pattern_mask(
+                                label_audio_ids.unsqueeze(0),
+                                bos_token_id=-100,
+                                pad_token_id=-100,
+                            )[0].squeeze(0)
+                new_audio_out_ids_l.append(audio_codes)
+
+                if return_labels:
+                    if audio_out_no_train_flag[idx]:
+                        label_audio_ids[:] = -100
+                    label_audio_ids_l.append(label_audio_ids)
+
+            audio_out_ids = torch.cat(new_audio_out_ids_l, dim=1).long()
+            if return_labels:
+                label_audio_ids = torch.cat(label_audio_ids_l, dim=1).long()
+            audio_out_ids_start = torch.cumsum(
+                torch.tensor([0] + [audio_codes.shape[1] for audio_codes in new_audio_out_ids_l[:-1]]), dim=0
+            )
+            audio_out_ids_start_group_loc = torch.tensor(audio_out_ids_group_loc_l, dtype=torch.long)
+        else:
+            audio_out_ids = torch.zeros((0, 0), dtype=torch.long)
+            audio_out_ids_start = torch.zeros(0, dtype=torch.long)
+            if return_labels:
+                label_audio_ids = torch.zeros((0, 0), dtype=torch.long)
+
+        reward = torch.tensor(reward_l, dtype=torch.float32)
+
+        # Handle padding for input ids and attention mask
+        if self.pad_left:
+            input_ids = torch.stack(
+                [
+                    F.pad(ele.input_ids, (max_seq_length - len(ele.input_ids), 0), value=self.pad_token_id)
+                    for ele in processed_batch
+                ]
+            )
+            if return_labels:
+                label_ids = torch.stack(
+                    [
+                        F.pad(ele.label_ids, (max_seq_length - len(ele.label_ids), 0), value=-100)
+                        for ele in processed_batch
+                    ]
+                )
+            attention_mask = torch.stack(
+                [
+                    F.pad(torch.ones_like(ele.input_ids), (max_seq_length - len(ele.input_ids), 0), value=0)
+                    for ele in processed_batch
+                ]
+            )
+        else:
+            input_ids = torch.stack(
+                [
+                    F.pad(ele.input_ids, (0, max_seq_length - len(ele.input_ids)), value=self.pad_token_id)
+                    for ele in processed_batch
+                ]
+            )
+            if return_labels:
+                label_ids = torch.stack(
+                    [
+                        F.pad(ele.label_ids, (0, max_seq_length - len(ele.label_ids)), value=-100)
+                        for ele in processed_batch
+                    ]
+                )
+            attention_mask = torch.stack(
+                [
+                    F.pad(torch.ones_like(ele.input_ids), (0, max_seq_length - len(ele.input_ids)), value=0)
+                    for ele in processed_batch
+                ]
+            )
+
+        if not self.return_audio_in_tokens:
+            audio_in_ids = None
+            audio_in_ids_start = None
+
+        # Apply audio_num_codebooks limit if specified
+        if self.audio_num_codebooks is not None:
+            if audio_in_ids is not None:
+                audio_in_ids = audio_in_ids[: self.audio_num_codebooks]
+            if audio_out_ids is not None:
+                audio_out_ids = audio_out_ids[: self.audio_num_codebooks]
+            if label_audio_ids is not None:
+                label_audio_ids = label_audio_ids[: self.audio_num_codebooks]
+
+        return HiggsAudioBatchInput(
+            input_ids=input_ids,
+            attention_mask=attention_mask,
+            audio_features=audio_features,
+            audio_feature_attention_mask=audio_feature_attention_mask,
+            audio_out_ids=audio_out_ids,
+            audio_out_ids_start=audio_out_ids_start,
+            audio_out_ids_start_group_loc=audio_out_ids_start_group_loc,
+            audio_in_ids=audio_in_ids,
+            audio_in_ids_start=audio_in_ids_start,
+            label_ids=label_ids,
+            label_audio_ids=label_audio_ids,
+            reward=reward,
+        )

train-higgs-audio/boson_multimodal/data_types.py

@@ -0,0 +1,38 @@
+"""Basic data types for multimodal ChatML format."""
+
+from dataclasses import dataclass
+from typing import Dict, List, Optional, Union
+
+
+@dataclass
+class AudioContent:
+    audio_url: str
+    # Base64 encoded audio bytes
+    raw_audio: Optional[str] = None
+    offset: Optional[float] = None
+    duration: Optional[float] = None
+    row_id: Optional[int] = None
+    type: str = "audio"
+
+
+@dataclass
+class TextContent:
+    text: str
+    type: str = "text"
+
+
+@dataclass
+class Message:
+    role: str
+    content: Union[str, AudioContent, TextContent, List[Union[str, AudioContent, TextContent]]]
+    recipient: Optional[str] = None
+
+
+@dataclass
+class ChatMLSample:
+    """Dataclass to hold multimodal ChatML data."""
+
+    messages: List[Message]
+    start_index: Optional[int] = None  # We will mask the messages[:start_index] when finetuning the LLM.
+    misc: Optional[Dict] = None
+    speaker: Optional[str] = None

train-higgs-audio/boson_multimodal/dataset/chatml_dataset copy.py

@@ -0,0 +1,533 @@
+import dacite
+import pandas as pd
+import torch
+import json
+
+import numpy as np
+import multiprocessing as mp
+
+from dataclasses import dataclass, fields
+from abc import ABC, abstractmethod
+from typing import Union, List, Dict, Optional
+
+from ..data_types import ChatMLSample, TextContent, AudioContent
+from ..constants import AUDIO_IN_TOKEN, AUDIO_OUT_TOKEN
+
+from loguru import logger
+
+# Whisper processor, 30 sec -> 3000 features
+# Then we divide 4 in the audio towker, we decrease 3000 features to 750, which gives 25 Hz
+WHISPER_EMBED_NUM_HIDDEN_STATE_PER_SEC = 25
+
+
+@dataclass
+class ChatMLDatasetSample:
+    input_ids: torch.LongTensor  # Shape (seq_len,): The input text tokens.
+    label_ids: torch.LongTensor  # Shape (seq_len,): The label ids.
+    audio_ids_concat: torch.LongTensor  # Shape (num_codebooks, audio_seq_len): The audio tokens that are concatenated.
+    # Here `audio_seq_len` is the length of the concatenated audio tokens.`
+    audio_ids_start: (
+        torch.LongTensor
+    )  # Shape (num_audios,): The start index of each audio token in the concatenated audio tokens.
+    audio_waveforms_concat: (
+        torch.Tensor
+    )  # Shape (total_wv_length,): The concatenated audio waveforms for audio-in features.
+    audio_waveforms_start: (
+        torch.LongTensor
+    )  # Shape (num_audios,): The start index of each audio waveform in the concatenated audio waveforms.
+    audio_sample_rate: torch.Tensor  # Shape (num_audios,): The sampling rate of the audio waveforms.
+    audio_speaker_indices: (
+        torch.LongTensor
+    )  # Shape (num_audios,) -1 means unknown speaker: The speaker indices for each audio.
+    audio_label_ids_concat: Optional[torch.LongTensor] = (
+        None  # Shape (num_codebooks, audio_seq_len): The audio tokens that are concatenated.
+    )
+    # Here `audio_seq_len` is the length of the concatenated audio tokens.`
+    reward: Optional[float] = None
+
+    def num_audios(self):
+        return max(len(self.audio_waveforms_start), len(self.audio_ids_start))
+
+    def get_audio_codes(self, idx):
+        code_start = self.audio_ids_start[idx]
+        if idx < len(self.audio_ids_start) - 1:
+            code_end = self.audio_ids_start[idx + 1]
+        else:
+            code_end = self.audio_ids_concat.shape[-1]
+
+        return self.audio_ids_concat[:, code_start:code_end]
+
+    def get_audio_codes_labels(self, idx):
+        if self.audio_label_ids_concat is None:
+            return None
+        code_start = self.audio_ids_start[idx]
+        if idx < len(self.audio_ids_start) - 1:
+            code_end = self.audio_ids_start[idx + 1]
+        else:
+            code_end = self.audio_ids_concat.shape[-1]
+
+        return self.audio_label_ids_concat[:, code_start:code_end]
+
+    def get_wv(self, idx):
+        wv_start = self.audio_waveforms_start[idx]
+        sr = self.audio_sample_rate[idx]
+        if idx < len(self.audio_waveforms_start) - 1:
+            wv_end = self.audio_waveforms_start[idx + 1]
+        else:
+            wv_end = self.audio_waveforms_concat.shape[-1]
+        return self.audio_waveforms_concat[wv_start:wv_end], sr
+
+    def cal_num_tokens(
+        self,
+        encode_whisper_embed: bool = True,
+        encode_audio_in_tokens: bool = False,
+        encode_audio_out_tokens: bool = True,
+        audio_in_token_id: int = 128015,
+        audio_out_token_id: int = 128016,
+    ) -> int:
+        # we firstly exclude <|AUDIO|> and <|AUDIO_OUT|> because we do late merging and replace those position with actual audio features and audio token ids
+        # It's assumed that we always have audio_ids when audio_waveforms are there (but not vice-versa)
+        num_tokens = len(self.input_ids) - len(self.audio_ids_start)
+
+        if encode_whisper_embed and len(self.audio_waveforms_concat) > 0:
+            audio_lengths = torch.diff(self.audio_waveforms_start)
+            if len(audio_lengths):
+                # Sum before calling .item()
+                num_tokens += (
+                    (
+                        np.ceil(WHISPER_EMBED_NUM_HIDDEN_STATE_PER_SEC * audio_lengths / self.audio_sample_rate[:-1])
+                    ).sum()
+                ).item()
+            # add the last audio's token estimation
+            num_tokens += (
+                np.ceil(
+                    WHISPER_EMBED_NUM_HIDDEN_STATE_PER_SEC
+                    * (self.audio_waveforms_concat.shape[0] - self.audio_waveforms_start[-1])
+                    / self.audio_sample_rate[-1]
+                )
+            ).item()
+
+        if self.audio_ids_concat.size(1) > 0:
+            audio_io_ids = self.input_ids[
+                (self.input_ids == audio_in_token_id) | (self.input_ids == audio_out_token_id)
+            ]
+            audio_io_id_lengths = torch.concat(
+                [
+                    torch.diff(self.audio_ids_start),
+                    torch.tensor([self.audio_ids_concat.shape[-1] - self.audio_ids_start[-1]]),
+                ]
+            )
+            if encode_audio_in_tokens:
+                num_tokens += torch.sum(audio_io_id_lengths[audio_io_ids == audio_in_token_id]).item()
+
+            if encode_audio_out_tokens:
+                num_tokens += torch.sum(audio_io_id_lengths[audio_io_ids == audio_out_token_id]).item()
+
+        return int(num_tokens)
+
+    @classmethod
+    def merge(
+        cls,
+        samples: List["ChatMLDatasetSample"],
+        eos_token_id: int,
+        ignore_index: int,
+        padding_size: Optional[int] = None,
+    ) -> "ChatMLDatasetSample":
+        """Merges a list of ChatMLDatasetSample instances, inserting eos_token_id and ignore_index between them, and adjusting offsets for audio_ids_start and audio_waveforms_start.
+
+        Args:
+            samples (List[ChatMLDatasetSample]): List of samples to merge.
+            eos_token_id (int): Tokens to be inserted into input_ids between samples.
+            ignore_index (int): Default label for padding.
+            padding_size (Optional[int]): If provided, pad the sequence to with this length.
+
+        Returns:
+            ChatMLDatasetSample: Merged and potentially padded sample.
+        """
+        if not samples:
+            logger.fatal("The samples list is empty and cannot be merged.")
+            raise ValueError("The samples list is empty and cannot be merged.")
+
+        # Initialize empty lists for concatenation
+        input_ids_list = []
+        label_ids_list = []
+        audio_ids_concat_list = []
+        audio_ids_start_list = []
+        audio_waveforms_concat_list = []
+        audio_waveforms_start_list = []
+        audio_sample_rate_list = []
+        audio_speaker_indices_list = []
+
+        # Track offsets
+        audio_ids_offset = 0
+        audio_waveforms_offset = 0
+
+        for sample in samples:
+            # Add input_ids and label_ids with padding
+            if input_ids_list:
+                input_ids_list.append(torch.tensor([eos_token_id], dtype=torch.long))
+                label_ids_list.append(torch.tensor([ignore_index], dtype=torch.long))
+            input_ids_list.append(sample.input_ids)
+            label_ids_list.append(sample.label_ids)
+
+            # Add audio_ids_concat and handle empty audio ids
+            if sample.audio_ids_concat.size(1) > 0:
+                audio_ids_concat_list.append(sample.audio_ids_concat)
+
+                # Offset and add audio_ids_start
+                audio_ids_start_list.append(sample.audio_ids_start + audio_ids_offset)
+                audio_ids_offset += sample.audio_ids_concat.size(
+                    1
+                )  # (num_codebooks, seq_len): Update offset by audio_seq_len
+
+            # Add audio_waveforms_concat
+            if sample.audio_waveforms_concat.size(0) > 0:
+                # Check dimensions of the audio waveform to ensure consistency
+                if (
+                    audio_waveforms_concat_list
+                    and sample.audio_waveforms_concat.dim() != audio_waveforms_concat_list[0].dim()
+                ):
+                    logger.warning(
+                        f"Skipping audio waveform with inconsistent dimensions: expected {audio_waveforms_concat_list[0].dim()}D, got {sample.audio_waveforms_concat.dim()}D"
+                    )
+                    continue
+
+                audio_waveforms_concat_list.append(sample.audio_waveforms_concat)
+                audio_waveforms_start_list.append(sample.audio_waveforms_start + audio_waveforms_offset)
+                audio_waveforms_offset += sample.audio_waveforms_concat.size(0)
+
+                # Add audio_sample_rate and audio_speaker_indices
+                audio_sample_rate_list.append(sample.audio_sample_rate)
+
+            audio_speaker_indices_list.append(sample.audio_speaker_indices)
+
+        # Concatenate all tensors
+        input_ids = torch.cat(input_ids_list, dim=0)
+        label_ids = torch.cat(label_ids_list, dim=0)
+
+        # Apply padding if padding_size is specified
+        if padding_size is not None and padding_size > 0:
+            input_ids = torch.cat([input_ids, torch.full((padding_size,), eos_token_id, dtype=torch.long)], dim=0)
+            label_ids = torch.cat([label_ids, torch.full((padding_size,), ignore_index, dtype=torch.long)], dim=0)
+
+        # Safely concatenate audio tensors with proper error handling
+        try:
+            audio_ids_concat = torch.cat(audio_ids_concat_list, dim=1) if audio_ids_concat_list else torch.tensor([[]])
+            audio_ids_start = torch.cat(audio_ids_start_list, dim=0) if audio_ids_start_list else torch.tensor([])
+
+            # Check for dimensional consistency in audio waveforms
+            if audio_waveforms_concat_list:
+                dims = [t.dim() for t in audio_waveforms_concat_list]
+                if not all(d == dims[0] for d in dims):
+                    # If dimensions don't match, log warning and filter out the problematic tensors
+                    logger.warning(
+                        f"Inconsistent dimensions in audio waveforms: {dims}. Filtering to keep only consistent ones."
+                    )
+                    expected_dim = max(set(dims), key=dims.count)  # Most common dimension
+                    audio_waveforms_concat_list = [t for t in audio_waveforms_concat_list if t.dim() == expected_dim]
+
+                    # Recalculate audio_waveforms_start with the filtered list
+                    if audio_waveforms_concat_list:
+                        audio_waveforms_offset = 0
+                        audio_waveforms_start_list = []
+                        for waveform in audio_waveforms_concat_list:
+                            audio_waveforms_start_list.append(torch.tensor([audio_waveforms_offset]))
+                            audio_waveforms_offset += waveform.size(0)
+
+            audio_waveforms_concat = (
+                torch.cat(audio_waveforms_concat_list, dim=0) if audio_waveforms_concat_list else torch.tensor([])
+            )
+            audio_waveforms_start = (
+                torch.cat(audio_waveforms_start_list, dim=0) if audio_waveforms_start_list else torch.tensor([])
+            )
+            audio_sample_rate = (
+                torch.cat(audio_sample_rate_list, dim=0) if audio_sample_rate_list else torch.tensor([])
+            )
+            audio_speaker_indices = (
+                torch.cat(audio_speaker_indices_list, dim=0) if audio_speaker_indices_list else torch.tensor([])
+            )
+
+        except RuntimeError as e:
+            logger.error(f"Error during tensor concatenation: {str(e)}")
+            logger.warning("Falling back to empty audio tensors")
+            # Fall back to empty tensors
+            audio_ids_concat = torch.tensor([[]])
+            audio_ids_start = torch.tensor([])
+            audio_waveforms_concat = torch.tensor([])
+            audio_waveforms_start = torch.tensor([])
+            audio_sample_rate = torch.tensor([])
+            audio_speaker_indices = torch.tensor([])
+
+        # Create the merged sample
+        merged_sample = cls(
+            input_ids=input_ids,
+            label_ids=label_ids,
+            audio_ids_concat=audio_ids_concat,
+            audio_ids_start=audio_ids_start,
+            audio_waveforms_concat=audio_waveforms_concat,
+            audio_waveforms_start=audio_waveforms_start,
+            audio_sample_rate=audio_sample_rate,
+            audio_speaker_indices=audio_speaker_indices,
+        )
+
+        return merged_sample
+
+
+@dataclass
+class RankedChatMLDatasetSampleTuple:
+    samples: List[ChatMLDatasetSample]
+    scores: List[float]
+
+    def max_score_sample(self) -> ChatMLDatasetSample:
+        idx = self.scores.index(max(self.scores))
+        self.samples[idx].reward = self.scores[idx]
+        return self.samples[idx]
+
+    def min_score_sample(self) -> ChatMLDatasetSample:
+        idx = self.scores.index(min(self.scores))
+        self.samples[idx].reward = self.scores[idx]
+        return self.samples[idx]
+
+
+@dataclass
+class ChatMLDatasetStorageSample:
+    input_tokens: torch.LongTensor
+    label_tokens: torch.LongTensor
+    audio_bytes_cache_dir_index: int
+    audio_codes_cache_dir_index: int
+    audio_bytes_indices: torch.LongTensor
+    audio_codes_indices: torch.LongTensor
+    speaker_indices: torch.LongTensor
+    file_index: int
+    original_sample_index: int
+
+
+# TODO(sxjscience): We need to revist the logic about parsing speaker ids.
+# Currently, we assume that the speaker id is stored at the "misc" field in ChatMLSample.
+def prepare_chatml_sample(sample: Union[ChatMLSample, Dict], tokenizer):
+    """Preprocess the ChatML sample to get the tokens for the text part.
+
+    Args:
+        sample (ChatMLSample): The ChatML sample to preprocess.
+        tokenizer: The tokenizer to use for encoding the text.
+
+    """
+
+    try:
+        if not isinstance(sample, ChatMLSample):
+            # Handle all fields that could be NaN
+            if "speaker" in sample and pd.isna(sample["speaker"]):
+                sample["speaker"] = None
+            if "start_index" in sample and pd.isna(sample["start_index"]):
+                sample["start_index"] = None
+            if "content" in sample and pd.isna(sample["content"]):
+                sample["content"] = ""
+
+            # Convert any other potential NaN values in nested structures
+            def convert_nan_to_none(obj):
+                import numpy as np
+
+                if isinstance(obj, (pd.Series, np.ndarray)):
+                    return obj.tolist()
+                elif pd.api.types.is_scalar(obj) and pd.isna(obj):
+                    return None
+                elif isinstance(obj, dict):
+                    return {k: convert_nan_to_none(v) for k, v in obj.items()}
+                elif isinstance(obj, (list, tuple)):  # Fixed: Handle both list and tuple
+                    return [convert_nan_to_none(item) for item in obj]
+                return obj
+
+            # Clean the sample data
+            clean_sample = convert_nan_to_none(sample)
+
+            val_keys = []
+            for field in fields(ChatMLSample):
+                if field.name in clean_sample:
+                    val_keys.append(field.name)
+            clean_sample = {k: clean_sample[k] for k in val_keys}
+
+            try:
+                sample = dacite.from_dict(
+                    data_class=ChatMLSample, data=clean_sample, config=dacite.Config(strict=True, check_types=True)
+                )
+            except Exception as e:
+                print(f"Failed to convert to ChatMLSample: {e}")
+                print(f"Clean sample: {json.dumps(clean_sample, indent=2)}")
+                return None, None, None, None
+
+        input_tokens = []
+        label_tokens = []
+        audio_contents = []
+        speaker_id = None
+        if sample.speaker is not None:
+            speaker_id = sample.speaker
+        elif sample.misc is not None:
+            if "speaker" in sample.misc:
+                speaker_id = sample.misc["speaker"]
+
+        total_m = len(sample.messages)
+        for turn_id, message in enumerate(sample.messages):
+            role = message.role
+            recipient = message.recipient
+            content = message.content
+            content_l = []
+
+            if isinstance(content, str):
+                content_l.append(TextContent(text=content))
+            elif isinstance(content, TextContent):
+                content_l.append(content)
+            elif isinstance(content, AudioContent):
+                content_l.append(content)
+            elif isinstance(content, list):
+                for ele in content:
+                    if isinstance(ele, str):
+                        content_l.append(TextContent(text=ele))
+                    else:
+                        content_l.append(ele)
+            if turn_id == 0:
+                prefix = f"<|begin_of_text|><|start_header_id|>{role}<|end_header_id|>\n\n"
+            else:
+                prefix = f"<|start_header_id|>{role}<|end_header_id|>\n\n"
+            eot_postfix = "<|eot_id|>"
+            eom_postfix = "<|eom_id|>"
+
+            prefix_tokens = tokenizer.encode(prefix, add_special_tokens=False)
+            input_tokens.extend(prefix_tokens)
+            label_tokens.extend([-100 for _ in prefix_tokens])
+
+            if recipient:
+                assert role == "assistant", "Recipient is only available for assistant role."
+                recipient_tokens = tokenizer.encode(f"{recipient}<|recipient|>", add_special_tokens=False)
+                input_tokens.extend(recipient_tokens)
+                label_tokens.extend(recipient_tokens)
+
+            for content in content_l:
+                if content.type == "text":
+                    text_tokens = tokenizer.encode(content.text, add_special_tokens=False)
+                    input_tokens.extend(text_tokens)
+                    if role == "assistant" and (sample.start_index is None or turn_id >= sample.start_index):
+                        label_tokens.extend(text_tokens)
+                    else:
+                        label_tokens.extend([-100 for _ in text_tokens])
+
+                elif content.type == "audio":
+                    # Generate the text-part of the audio tokens
+                    audio_contents.append(content)
+                    if role == "user" or role == "system":
+                        # Add the text tokens
+                        text_tokens = tokenizer.encode(
+                            f"<|audio_bos|><|AUDIO|><|audio_eos|>",
+                            add_special_tokens=False,
+                        )
+                        input_tokens.extend(text_tokens)
+                        label_tokens.extend([-100 for _ in text_tokens])
+                    elif role == "assistant":
+                        # Add the text tokens for audio-out part.
+                        text_tokens = tokenizer.encode(
+                            f"<|audio_out_bos|><|AUDIO_OUT|><|audio_eos|>",
+                            add_special_tokens=False,
+                        )
+                        input_tokens.extend(text_tokens)
+                        if sample.start_index is None or turn_id >= sample.start_index:
+                            label_tokens.extend(text_tokens)
+                        else:
+                            label_tokens.extend([-100 for _ in text_tokens])
+            next_id = turn_id + 1
+            if role == "assistant" and next_id != total_m and sample.messages[next_id].role == "assistant":
+                postfix_tokens = tokenizer.encode(eom_postfix, add_special_tokens=False)
+                input_tokens.extend(postfix_tokens)
+            else:
+                postfix_tokens = tokenizer.encode(eot_postfix, add_special_tokens=False)
+                input_tokens.extend(postfix_tokens)
+            if role == "assistant" and (sample.start_index is None or turn_id >= sample.start_index):
+                label_tokens.extend(postfix_tokens)
+            else:
+                label_tokens.extend([-100 for _ in postfix_tokens])
+
+        return input_tokens, label_tokens, audio_contents, speaker_id
+
+    except Exception as e:
+        print(f"Error in prepare_chatml_sample: {str(e)}")
+        print(f"Sample data: {json.dumps(sample, indent=2)}")
+        return None, None, None, None
+
+
+def extract_generation_prompt_from_input_tokens(input_tokens, tokenizer):
+    """Extract the generation prompt and reference answer from the input tokens.
+
+    For example:
+
+    Input Text = '<|begin_of_text|><|start_header_id|>user<|end_header_id|>\n\n
+    What words do you hear from the provided audio? Write it down for me.<|audio_bos|><|AUDIO|><|audio_eos|><|eot_id|>
+    <|start_header_id|>assistant<|end_header_id|>\n\nAt first they went by quick, too quick to even get.<|eot_id|>'
+
+    -->
+
+    Prompt = '<|begin_of_text|><|start_header_id|>user<|end_header_id|>\n\n
+    What words do you hear from the provided audio? Write it down for me.<|audio_bos|><|AUDIO|><|audio_eos|><|eot_id|>
+    <|start_header_id|>assistant<|end_header_id|>\n\n',
+    Reference = 'At first they went by quick, too quick to even get.'
+
+    Args:
+        input_tokens: The input tokens.
+        audio_contents: The audio contents.
+        tokenizer: The tokenizer to use for decoding the text.
+
+    Returns:
+        prompt_tokens: The tokens for the prompt.
+        reference_answer: The reference answer.
+        num_audios_in_reference: The number of audios in the reference answer.
+
+    """
+    input_text = tokenizer.decode(input_tokens)
+    generation_prefix = "<|start_header_id|>assistant<|end_header_id|>\n\n"
+    postfix = "<|eot_id|>"
+    assert generation_prefix in input_text
+    generation_prompt_end_loc = input_text.rfind(generation_prefix) + len(generation_prefix)
+    generation_prompt = input_text[:generation_prompt_end_loc]
+    reference_answer = input_text[generation_prompt_end_loc : input_text.find(postfix, generation_prompt_end_loc)]
+    num_audios_in_reference = reference_answer.count(AUDIO_IN_TOKEN) + reference_answer.count(AUDIO_OUT_TOKEN)
+    return tokenizer.encode(generation_prompt, add_special_tokens=False), reference_answer, num_audios_in_reference
+
+
+def prepare_chatml_dataframe_single_process(df, tokenizer):
+    """Prepare the ChatML DataFrame."""
+    ret = []
+    for _, row in df.iterrows():
+        input_tokens, label_tokens, audio_contents, speaker_id = prepare_chatml_sample(row.to_dict(), tokenizer)
+        ret.append((input_tokens, label_tokens, audio_contents, speaker_id))
+    return ret
+
+
+def prepare_chatml_dataframe(df, tokenizer, num_process=16):
+    if num_process is None:
+        return prepare_chatml_dataframe_single_process(df, tokenizer)
+    else:
+        num_process = max(min(len(df) // 1000, num_process), 1)
+        workloads = np.array_split(df, num_process)
+        with mp.Pool(num_process) as pool:
+            ret = pool.starmap(
+                prepare_chatml_dataframe_single_process, [(workload, tokenizer) for workload in workloads]
+            )
+    return sum(ret, [])
+
+
+class DatasetInterface(ABC):
+    @abstractmethod
+    def __getitem__(self, idx) -> Union["ChatMLDatasetSample", "RankedChatMLDatasetSampleTuple"]:
+        """Retrieve a dataset sample by index."""
+        raise NotImplementedError
+
+
+class IterableDatasetInterface(ABC):
+    @abstractmethod
+    def __iter__(self) -> Union["ChatMLDatasetSample", "RankedChatMLDatasetSampleTuple"]:
+        """Retrieve a sample by iterating through the dataset."""
+        raise NotImplementedError
+
+
+@dataclass
+class DatasetInfo:
+    dataset_type: str
+    group_type: Optional[str] = None
+    mask_text: Optional[bool] = None  # Whether to mask the text tokens for pretraining samples.

train-higgs-audio/boson_multimodal/dataset/chatml_dataset.py

@@ -0,0 +1,655 @@
+import dacite
+import pandas as pd
+import torch
+import json
+
+import numpy as np
+import multiprocessing as mp
+
+from dataclasses import dataclass, fields
+from abc import ABC, abstractmethod
+from typing import Union, List, Dict, Optional
+
+from ..data_types import ChatMLSample, TextContent, AudioContent
+from ..constants import AUDIO_IN_TOKEN, AUDIO_OUT_TOKEN
+
+from loguru import logger
+
+# Whisper processor, 30 sec -> 3000 features
+# Then we divide 4 in the audio towker, we decrease 3000 features to 750, which gives 25 Hz
+WHISPER_EMBED_NUM_HIDDEN_STATE_PER_SEC = 25
+
+
+@dataclass
+class ChatMLDatasetSample:
+    input_ids: torch.LongTensor  # Shape (seq_len,): The input text tokens.
+    label_ids: torch.LongTensor  # Shape (seq_len,): The label ids.
+    audio_ids_concat: torch.LongTensor  # Shape (num_codebooks, audio_seq_len): The audio tokens that are concatenated.
+    # Here `audio_seq_len` is the length of the concatenated audio tokens.`
+    audio_ids_start: (
+        torch.LongTensor
+    )  # Shape (num_audios,): The start index of each audio token in the concatenated audio tokens.
+    audio_waveforms_concat: (
+        torch.Tensor
+    )  # Shape (total_wv_length,): The concatenated audio waveforms for audio-in features.
+    audio_waveforms_start: (
+        torch.LongTensor
+    )  # Shape (num_audios,): The start index of each audio waveform in the concatenated audio waveforms.
+    audio_sample_rate: torch.Tensor  # Shape (num_audios,): The sampling rate of the audio waveforms.
+    audio_speaker_indices: (
+        torch.LongTensor
+    )  # Shape (num_audios,) -1 means unknown speaker: The speaker indices for each audio.
+    audio_label_ids_concat: Optional[torch.LongTensor] = (
+        None  # Shape (num_codebooks, audio_seq_len): The audio tokens that are concatenated.
+    )
+    # Here `audio_seq_len` is the length of the concatenated audio tokens.`
+    label_audio_ids: Optional[torch.LongTensor] = None  # 新增字段
+    reward: Optional[float] = None
+
+    def num_audios(self):
+        return max(len(self.audio_waveforms_start), len(self.audio_ids_start))
+
+    def get_audio_codes(self, idx):
+        code_start = self.audio_ids_start[idx]
+        if idx < len(self.audio_ids_start) - 1:
+            code_end = self.audio_ids_start[idx + 1]
+        else:
+            code_end = self.audio_ids_concat.shape[-1]
+
+        return self.audio_ids_concat[:, code_start:code_end]
+
+    def get_audio_codes_labels(self, idx):
+        if self.audio_label_ids_concat is None:
+            return None
+        code_start = self.audio_ids_start[idx]
+        if idx < len(self.audio_ids_start) - 1:
+            code_end = self.audio_ids_start[idx + 1]
+        else:
+            code_end = self.audio_ids_concat.shape[-1]
+
+        return self.audio_label_ids_concat[:, code_start:code_end]
+
+    def get_wv(self, idx):
+        wv_start = self.audio_waveforms_start[idx]
+        sr = self.audio_sample_rate[idx]
+        if idx < len(self.audio_waveforms_start) - 1:
+            wv_end = self.audio_waveforms_start[idx + 1]
+        else:
+            wv_end = self.audio_waveforms_concat.shape[-1]
+        return self.audio_waveforms_concat[wv_start:wv_end], sr
+
+    def cal_num_tokens(
+        self,
+        encode_whisper_embed: bool = True,
+        encode_audio_in_tokens: bool = False,
+        encode_audio_out_tokens: bool = True,
+        audio_in_token_id: int = 128015,
+        audio_out_token_id: int = 128016,
+    ) -> int:
+        # we firstly exclude <|AUDIO|> and <|AUDIO_OUT|> because we do late merging and replace those position with actual audio features and audio token ids
+        # It's assumed that we always have audio_ids when audio_waveforms are there (but not vice-versa)
+        num_tokens = len(self.input_ids) - len(self.audio_ids_start)
+
+        if encode_whisper_embed and len(self.audio_waveforms_concat) > 0:
+            audio_lengths = torch.diff(self.audio_waveforms_start)
+            if len(audio_lengths):
+                # Sum before calling .item()
+                num_tokens += (
+                    (
+                        np.ceil(WHISPER_EMBED_NUM_HIDDEN_STATE_PER_SEC * audio_lengths / self.audio_sample_rate[:-1])
+                    ).sum()
+                ).item()
+            # add the last audio's token estimation
+            num_tokens += (
+                np.ceil(
+                    WHISPER_EMBED_NUM_HIDDEN_STATE_PER_SEC
+                    * (self.audio_waveforms_concat.shape[0] - self.audio_waveforms_start[-1])
+                    / self.audio_sample_rate[-1]
+                )
+            ).item()
+
+        if self.audio_ids_concat.size(1) > 0:
+            audio_io_ids = self.input_ids[
+                (self.input_ids == audio_in_token_id) | (self.input_ids == audio_out_token_id)
+            ]
+            audio_io_id_lengths = torch.concat(
+                [
+                    torch.diff(self.audio_ids_start),
+                    torch.tensor([self.audio_ids_concat.shape[-1] - self.audio_ids_start[-1]]),
+                ]
+            )
+            if encode_audio_in_tokens:
+                num_tokens += torch.sum(audio_io_id_lengths[audio_io_ids == audio_in_token_id]).item()
+
+            if encode_audio_out_tokens:
+                num_tokens += torch.sum(audio_io_id_lengths[audio_io_ids == audio_out_token_id]).item()
+
+        return int(num_tokens)
+
+    @classmethod
+    def merge(
+        cls,
+        samples: List["ChatMLDatasetSample"],
+        eos_token_id: int,
+        ignore_index: int,
+        padding_size: Optional[int] = None,
+    ) -> "ChatMLDatasetSample":
+        """Merges a list of ChatMLDatasetSample instances, inserting eos_token_id and ignore_index between them, and adjusting offsets for audio_ids_start and audio_waveforms_start.
+
+        Args:
+            samples (List[ChatMLDatasetSample]): List of samples to merge.
+            eos_token_id (int): Tokens to be inserted into input_ids between samples.
+            ignore_index (int): Default label for padding.
+            padding_size (Optional[int]): If provided, pad the sequence to with this length.
+
+        Returns:
+            ChatMLDatasetSample: Merged and potentially padded sample.
+        """
+        if not samples:
+            logger.fatal("The samples list is empty and cannot be merged.")
+            raise ValueError("The samples list is empty and cannot be merged.")
+
+        # Initialize empty lists for concatenation
+        input_ids_list = []
+        label_ids_list = []
+        audio_ids_concat_list = []
+        audio_ids_start_list = []
+        audio_waveforms_concat_list = []
+        audio_waveforms_start_list = []
+        audio_sample_rate_list = []
+        audio_speaker_indices_list = []
+        label_audio_ids_list = []  # 新增
+
+        # Track offsets
+        audio_ids_offset = 0
+        audio_waveforms_offset = 0
+
+        for sample in samples:
+            # Add input_ids and label_ids with padding
+            if input_ids_list:
+                input_ids_list.append(torch.tensor([eos_token_id], dtype=torch.long))
+                label_ids_list.append(torch.tensor([ignore_index], dtype=torch.long))
+            input_ids_list.append(sample.input_ids)
+            label_ids_list.append(sample.label_ids)
+
+            # Add audio_ids_concat and handle empty audio ids
+            if sample.audio_ids_concat.size(1) > 0:
+                audio_ids_concat_list.append(sample.audio_ids_concat)
+
+                # Offset and add audio_ids_start
+                audio_ids_start_list.append(sample.audio_ids_start + audio_ids_offset)
+                audio_ids_offset += sample.audio_ids_concat.size(
+                    1
+                )  # (num_codebooks, seq_len): Update offset by audio_seq_len
+
+            # Add label_audio_ids
+            if sample.label_audio_ids is not None:
+                label_audio_ids_list.append(sample.label_audio_ids)
+
+            # Add audio_waveforms_concat
+            if sample.audio_waveforms_concat.size(0) > 0:
+                # Check dimensions of the audio waveform to ensure consistency
+                if (
+                    audio_waveforms_concat_list
+                    and sample.audio_waveforms_concat.dim() != audio_waveforms_concat_list[0].dim()
+                ):
+                    logger.warning(
+                        f"Skipping audio waveform with inconsistent dimensions: expected {audio_waveforms_concat_list[0].dim()}D, got {sample.audio_waveforms_concat.dim()}D"
+                    )
+                    continue
+
+                audio_waveforms_concat_list.append(sample.audio_waveforms_concat)
+                audio_waveforms_start_list.append(sample.audio_waveforms_start + audio_waveforms_offset)
+                audio_waveforms_offset += sample.audio_waveforms_concat.size(0)
+
+                # Add audio_sample_rate and audio_speaker_indices
+                audio_sample_rate_list.append(sample.audio_sample_rate)
+
+            audio_speaker_indices_list.append(sample.audio_speaker_indices)
+
+        # Concatenate all tensors
+        input_ids = torch.cat(input_ids_list, dim=0)
+        label_ids = torch.cat(label_ids_list, dim=0)
+
+        # Apply padding if padding_size is specified
+        if padding_size is not None and padding_size > 0:
+            input_ids = torch.cat([input_ids, torch.full((padding_size,), eos_token_id, dtype=torch.long)], dim=0)
+            label_ids = torch.cat([label_ids, torch.full((padding_size,), ignore_index, dtype=torch.long)], dim=0)
+
+        # Safely concatenate audio tensors with proper error handling
+        try:
+            audio_ids_concat = torch.cat(audio_ids_concat_list, dim=1) if audio_ids_concat_list else torch.tensor([[]])
+            audio_ids_start = torch.cat(audio_ids_start_list, dim=0) if audio_ids_start_list else torch.tensor([])
+            label_audio_ids = torch.cat(label_audio_ids_list, dim=-1) if label_audio_ids_list else None
+
+            # Check for dimensional consistency in audio waveforms
+            if audio_waveforms_concat_list:
+                dims = [t.dim() for t in audio_waveforms_concat_list]
+                if not all(d == dims[0] for d in dims):
+                    # If dimensions don't match, log warning and filter out the problematic tensors
+                    logger.warning(
+                        f"Inconsistent dimensions in audio waveforms: {dims}. Filtering to keep only consistent ones."
+                    )
+                    expected_dim = max(set(dims), key=dims.count)  # Most common dimension
+                    audio_waveforms_concat_list = [t for t in audio_waveforms_concat_list if t.dim() == expected_dim]
+
+                    # Recalculate audio_waveforms_start with the filtered list
+                    if audio_waveforms_concat_list:
+                        audio_waveforms_offset = 0
+                        audio_waveforms_start_list = []
+                        for waveform in audio_waveforms_concat_list:
+                            audio_waveforms_start_list.append(torch.tensor([audio_waveforms_offset]))
+                            audio_waveforms_offset += waveform.size(0)
+
+            audio_waveforms_concat = (
+                torch.cat(audio_waveforms_concat_list, dim=0) if audio_waveforms_concat_list else torch.tensor([])
+            )
+            audio_waveforms_start = (
+                torch.cat(audio_waveforms_start_list, dim=0) if audio_waveforms_start_list else torch.tensor([])
+            )
+            audio_sample_rate = (
+                torch.cat(audio_sample_rate_list, dim=0) if audio_sample_rate_list else torch.tensor([])
+            )
+            audio_speaker_indices = (
+                torch.cat(audio_speaker_indices_list, dim=0) if audio_speaker_indices_list else torch.tensor([])
+            )
+
+        except RuntimeError as e:
+            logger.error(f"Error during tensor concatenation: {str(e)}")
+            logger.warning("Falling back to empty audio tensors")
+            # Fall back to empty tensors
+            audio_ids_concat = torch.tensor([[]])
+            audio_ids_start = torch.tensor([])
+            audio_waveforms_concat = torch.tensor([])
+            audio_waveforms_start = torch.tensor([])
+            audio_sample_rate = torch.tensor([])
+            audio_speaker_indices = torch.tensor([])
+            label_audio_ids = None
+
+        # 在合并时也复制
+        label_audio_ids = audio_ids_concat if audio_ids_concat.numel() > 0 else None
+
+        merged_sample = cls(
+            input_ids=input_ids,
+            label_ids=label_ids,
+            audio_ids_concat=audio_ids_concat,
+            audio_ids_start=audio_ids_start,
+            audio_waveforms_concat=audio_waveforms_concat,
+            audio_waveforms_start=audio_waveforms_start,
+            audio_sample_rate=audio_sample_rate,
+            audio_speaker_indices=audio_speaker_indices,
+            audio_label_ids_concat=audio_label_ids_concat,  # 原来的字段
+            label_audio_ids=audio_ids_concat,  # 直接复制
+        )
+
+
+        return merged_sample
+
+
+@dataclass
+class RankedChatMLDatasetSampleTuple:
+    samples: List[ChatMLDatasetSample]
+    scores: List[float]
+
+    def max_score_sample(self) -> ChatMLDatasetSample:
+        idx = self.scores.index(max(self.scores))
+        self.samples[idx].reward = self.scores[idx]
+        return self.samples[idx]
+
+    def min_score_sample(self) -> ChatMLDatasetSample:
+        idx = self.scores.index(min(self.scores))
+        self.samples[idx].reward = self.scores[idx]
+        return self.samples[idx]
+
+
+@dataclass
+class ChatMLDatasetStorageSample:
+    input_tokens: torch.LongTensor
+    label_tokens: torch.LongTensor
+    audio_bytes_cache_dir_index: int
+    audio_codes_cache_dir_index: int
+    audio_bytes_indices: torch.LongTensor
+    audio_codes_indices: torch.LongTensor
+    speaker_indices: torch.LongTensor
+    file_index: int
+    original_sample_index: int
+
+
+def encode_audio_to_tokens(audio_content: AudioContent):
+    """
+    将音频内容编码为token IDs
+    这是一个示例函数，需要根据你的实际音频编码器实现
+    """
+    # 如果音频内容已经包含编码后的tokens
+    if hasattr(audio_content, 'codes') and audio_content.codes is not None:
+        if isinstance(audio_content.codes, torch.Tensor):
+            return audio_content.codes
+        else:
+            return torch.tensor(audio_content.codes, dtype=torch.long)
+    
+    # 如果没有预编码的tokens，返回空tensor
+    logger.warning(f"Cannot encode audio content: {audio_content}")
+    return torch.tensor([], dtype=torch.long)
+
+
+# TODO(sxjscience): We need to revist the logic about parsing speaker ids.
+# Currently, we assume that the speaker id is stored at the "misc" field in ChatMLSample.
+def prepare_chatml_sample(sample: Union[ChatMLSample, Dict], tokenizer):
+    """Preprocess the ChatML sample to get the tokens for the text part.
+
+    Args:
+        sample (ChatMLSample): The ChatML sample to preprocess.
+        tokenizer: The tokenizer to use for encoding the text.
+
+    """
+
+    try:
+        if not isinstance(sample, ChatMLSample):
+            # Handle all fields that could be NaN
+            if "speaker" in sample and pd.isna(sample["speaker"]):
+                sample["speaker"] = None
+            if "start_index" in sample and pd.isna(sample["start_index"]):
+                sample["start_index"] = None
+            if "content" in sample and pd.isna(sample["content"]):
+                sample["content"] = ""
+
+            # Convert any other potential NaN values in nested structures
+            def convert_nan_to_none(obj):
+                import numpy as np
+
+                if isinstance(obj, (pd.Series, np.ndarray)):
+                    return obj.tolist()
+                elif pd.api.types.is_scalar(obj) and pd.isna(obj):
+                    return None
+                elif isinstance(obj, dict):
+                    return {k: convert_nan_to_none(v) for k, v in obj.items()}
+                elif isinstance(obj, (list, tuple)):  # Fixed: Handle both list and tuple
+                    return [convert_nan_to_none(item) for item in obj]
+                return obj
+
+            # Clean the sample data
+            clean_sample = convert_nan_to_none(sample)
+
+            val_keys = []
+            for field in fields(ChatMLSample):
+                if field.name in clean_sample:
+                    val_keys.append(field.name)
+            clean_sample = {k: clean_sample[k] for k in val_keys}
+
+            try:
+                sample = dacite.from_dict(
+                    data_class=ChatMLSample, data=clean_sample, config=dacite.Config(strict=True, check_types=True)
+                )
+            except Exception as e:
+                print(f"Failed to convert to ChatMLSample: {e}")
+                print(f"Clean sample: {json.dumps(clean_sample, indent=2)}")
+                return None, None, None, None, None
+
+        input_tokens = []
+        label_tokens = []
+        audio_contents = []
+        audio_label_contents = []  # 新增：收集音频标签
+        speaker_id = None
+        if sample.speaker is not None:
+            speaker_id = sample.speaker
+        elif sample.misc is not None:
+            if "speaker" in sample.misc:
+                speaker_id = sample.misc["speaker"]
+
+        total_m = len(sample.messages)
+        for turn_id, message in enumerate(sample.messages):
+            role = message.role
+            recipient = message.recipient
+            content = message.content
+            content_l = []
+
+            if isinstance(content, str):
+                content_l.append(TextContent(text=content))
+            elif isinstance(content, TextContent):
+                content_l.append(content)
+            elif isinstance(content, AudioContent):
+                content_l.append(content)
+            elif isinstance(content, list):
+                for ele in content:
+                    if isinstance(ele, str):
+                        content_l.append(TextContent(text=ele))
+                    else:
+                        content_l.append(ele)
+            if turn_id == 0:
+                prefix = f"<|begin_of_text|><|start_header_id|>{role}<|end_header_id|>\n\n"
+            else:
+                prefix = f"<|start_header_id|>{role}<|end_header_id|>\n\n"
+            eot_postfix = "<|eot_id|>"
+            eom_postfix = "<|eom_id|>"
+
+            prefix_tokens = tokenizer.encode(prefix, add_special_tokens=False)
+            input_tokens.extend(prefix_tokens)
+            label_tokens.extend([-100 for _ in prefix_tokens])
+
+            if recipient:
+                assert role == "assistant", "Recipient is only available for assistant role."
+                recipient_tokens = tokenizer.encode(f"{recipient}<|recipient|>", add_special_tokens=False)
+                input_tokens.extend(recipient_tokens)
+                label_tokens.extend(recipient_tokens)
+
+            for content in content_l:
+                if content.type == "text":
+                    text_tokens = tokenizer.encode(content.text, add_special_tokens=False)
+                    input_tokens.extend(text_tokens)
+                    if role == "assistant" and (sample.start_index is None or turn_id >= sample.start_index):
+                        label_tokens.extend(text_tokens)
+                    else:
+                        label_tokens.extend([-100 for _ in text_tokens])
+
+                elif content.type == "audio":
+                    # 收集所有音频内容
+                    audio_contents.append(content)
+                    
+                    # 判断是否作为标签
+                    if role == "assistant" and (sample.start_index is None or turn_id >= sample.start_index):
+                        # assistant的音频输出作为标签
+                        audio_label_contents.append(content)
+                    else:
+                        # 其他情况不作为标签
+                        audio_label_contents.append(None)
+                    
+                    if role == "user" or role == "system":
+                        # Add the text tokens
+                        text_tokens = tokenizer.encode(
+                            f"<|audio_bos|><|AUDIO|><|audio_eos|>",
+                            add_special_tokens=False,
+                        )
+                        input_tokens.extend(text_tokens)
+                        label_tokens.extend([-100 for _ in text_tokens])
+                    elif role == "assistant":
+                        # Add the text tokens for audio-out part.
+                        text_tokens = tokenizer.encode(
+                            f"<|audio_out_bos|><|AUDIO_OUT|><|audio_eos|>",
+                            add_special_tokens=False,
+                        )
+                        input_tokens.extend(text_tokens)
+                        if sample.start_index is None or turn_id >= sample.start_index:
+                            label_tokens.extend(text_tokens)
+                        else:
+                            label_tokens.extend([-100 for _ in text_tokens])
+            next_id = turn_id + 1
+            if role == "assistant" and next_id != total_m and sample.messages[next_id].role == "assistant":
+                postfix_tokens = tokenizer.encode(eom_postfix, add_special_tokens=False)
+                input_tokens.extend(postfix_tokens)
+            else:
+                postfix_tokens = tokenizer.encode(eot_postfix, add_special_tokens=False)
+                input_tokens.extend(postfix_tokens)
+            if role == "assistant" and (sample.start_index is None or turn_id >= sample.start_index):
+                label_tokens.extend(postfix_tokens)
+            else:
+                label_tokens.extend([-100 for _ in postfix_tokens])
+
+        return input_tokens, label_tokens, audio_contents, audio_label_contents, speaker_id
+
+    except Exception as e:
+        print(f"Error in prepare_chatml_sample: {str(e)}")
+        print(f"Sample data: {json.dumps(sample, indent=2)}")
+        return None, None, None, None, None
+
+
+def extract_generation_prompt_from_input_tokens(input_tokens, tokenizer):
+    """Extract the generation prompt and reference answer from the input tokens.
+
+    For example:
+
+    Input Text = '<|begin_of_text|><|start_header_id|>user<|end_header_id|>\n\n
+    What words do you hear from the provided audio? Write it down for me.<|audio_bos|><|AUDIO|><|audio_eos|><|eot_id|>
+    <|start_header_id|>assistant<|end_header_id|>\n\nAt first they went by quick, too quick to even get.<|eot_id|>'
+
+    -->
+
+    Prompt = '<|begin_of_text|><|start_header_id|>user<|end_header_id|>\n\n
+    What words do you hear from the provided audio? Write it down for me.<|audio_bos|><|AUDIO|><|audio_eos|><|eot_id|>
+    <|start_header_id|>assistant<|end_header_id|>\n\n',
+    Reference = 'At first they went by quick, too quick to even get.'
+
+    Args:
+        input_tokens: The input tokens.
+        audio_contents: The audio contents.
+        tokenizer: The tokenizer to use for decoding the text.
+
+    Returns:
+        prompt_tokens: The tokens for the prompt.
+        reference_answer: The reference answer.
+        num_audios_in_reference: The number of audios in the reference answer.
+
+    """
+    input_text = tokenizer.decode(input_tokens)
+    generation_prefix = "<|start_header_id|>assistant<|end_header_id|>\n\n"
+    postfix = "<|eot_id|>"
+    assert generation_prefix in input_text
+    generation_prompt_end_loc = input_text.rfind(generation_prefix) + len(generation_prefix)
+    generation_prompt = input_text[:generation_prompt_end_loc]
+    reference_answer = input_text[generation_prompt_end_loc : input_text.find(postfix, generation_prompt_end_loc)]
+    num_audios_in_reference = reference_answer.count(AUDIO_IN_TOKEN) + reference_answer.count(AUDIO_OUT_TOKEN)
+    return tokenizer.encode(generation_prompt, add_special_tokens=False), reference_answer, num_audios_in_reference
+
+
+def prepare_chatml_dataframe_single_process(df, tokenizer):
+    """Prepare the ChatML DataFrame."""
+    ret = []
+    for _, row in df.iterrows():
+        result = prepare_chatml_sample(row.to_dict(), tokenizer)
+        if result[0] is not None:  # 检查是否成功处理
+            input_tokens, label_tokens, audio_contents, audio_label_contents, speaker_id = result
+            
+            # 处理音频数据
+            audio_ids_list = []
+            audio_waveforms_list = []
+            audio_sample_rates = []
+            audio_speaker_indices = []
+            audio_ids_start = []
+            audio_waveforms_start = []
+            label_audio_ids_list = []
+            
+            current_audio_offset = 0
+            current_waveform_offset = 0
+            
+            for i, (audio_content, audio_label_content) in enumerate(zip(audio_contents, audio_label_contents)):
+                # 编码音频为tokens
+                audio_tokens = encode_audio_to_tokens(audio_content)
+                
+                if audio_tokens.numel() > 0:
+                    # 记录起始位置
+                    audio_ids_start.append(current_audio_offset)
+                    
+                    # 添加音频tokens
+                    audio_ids_list.append(audio_tokens)
+                    current_audio_offset += audio_tokens.shape[-1]
+                
+                # 处理音频标签
+                if audio_label_content is not None:
+                    label_tokens = encode_audio_to_tokens(audio_label_content)
+                    label_audio_ids_list.append(label_tokens)
+                
+                # 处理音频波形
+                if hasattr(audio_content, 'waveform') and audio_content.waveform is not None:
+                    waveform = audio_content.waveform
+                    if isinstance(waveform, (list, np.ndarray)):
+                        waveform = torch.tensor(waveform, dtype=torch.float32)
+                    
+                    audio_waveforms_start.append(current_waveform_offset)
+                    audio_waveforms_list.append(waveform)
+                    current_waveform_offset += len(waveform)
+                    
+                    # 采样率
+                    sample_rate = getattr(audio_content, 'sample_rate', 16000)
+                    audio_sample_rates.append(sample_rate)
+                
+                # 说话人索引
+                speaker_idx = -1  # 默认未知说话人
+                if speaker_id is not None:
+                    speaker_idx = hash(speaker_id) % 1000  # 简单的hash映射
+                audio_speaker_indices.append(speaker_idx)
+            
+            # 拼接所有音频数据
+            if audio_ids_list:
+                audio_ids_concat = torch.cat(audio_ids_list, dim=-1)
+            else:
+                audio_ids_concat = torch.tensor([[]], dtype=torch.long)
+            
+            if audio_waveforms_list:
+                audio_waveforms_concat = torch.cat(audio_waveforms_list, dim=0)
+            else:
+                audio_waveforms_concat = torch.tensor([], dtype=torch.float32)
+            
+            if label_audio_ids_list:
+                label_audio_ids = torch.cat(label_audio_ids_list, dim=-1)
+            else:
+                label_audio_ids = None
+            
+            # 构建样本
+            sample = ChatMLDatasetSample(
+                input_ids=torch.tensor(input_tokens, dtype=torch.long),
+                label_ids=torch.tensor(label_tokens, dtype=torch.long),
+                audio_ids_concat=audio_ids_concat,
+                audio_ids_start=torch.tensor(audio_ids_start, dtype=torch.long),
+                audio_waveforms_concat=audio_waveforms_concat,
+                audio_waveforms_start=torch.tensor(audio_waveforms_start, dtype=torch.long),
+                audio_sample_rate=torch.tensor(audio_sample_rates, dtype=torch.float32),
+                audio_speaker_indices=torch.tensor(audio_speaker_indices, dtype=torch.long),
+                label_audio_ids=label_audio_ids,  # 设置音频标签
+            )
+            ret.append(sample)
+        else:
+            logger.warning(f"Failed to process sample at index {_}")
+    return ret
+
+
+def prepare_chatml_dataframe(df, tokenizer, num_process=16):
+    if num_process is None:
+        return prepare_chatml_dataframe_single_process(df, tokenizer)
+    else:
+        num_process = max(min(len(df) // 1000, num_process), 1)
+        workloads = np.array_split(df, num_process)
+        with mp.Pool(num_process) as pool:
+            ret = pool.starmap(
+                prepare_chatml_dataframe_single_process, [(workload, tokenizer) for workload in workloads]
+            )
+    return sum(ret, [])
+
+
+class DatasetInterface(ABC):
+    @abstractmethod
+    def __getitem__(self, idx) -> Union["ChatMLDatasetSample", "RankedChatMLDatasetSampleTuple"]:
+        """Retrieve a dataset sample by index."""
+        raise NotImplementedError
+
+
+class IterableDatasetInterface(ABC):
+    @abstractmethod
+    def __iter__(self) -> Union["ChatMLDatasetSample", "RankedChatMLDatasetSampleTuple"]:
+        """Retrieve a sample by iterating through the dataset."""
+        raise NotImplementedError
+
+
+@dataclass
+class DatasetInfo:
+    dataset_type: str
+    group_type: Optional[str] = None
+    mask_text: Optional[bool] = None  # Whether to mask the text tokens for pretraining samples.

train-higgs-audio/boson_multimodal/model/higgs_audio/__init__.py

@@ -0,0 +1,9 @@
+from transformers import AutoConfig, AutoModel
+
+from .configuration_higgs_audio import HiggsAudioConfig, HiggsAudioEncoderConfig
+from .modeling_higgs_audio import HiggsAudioModel
+
+
+AutoConfig.register("higgs_audio_encoder", HiggsAudioEncoderConfig)
+AutoConfig.register("higgs_audio", HiggsAudioConfig)
+AutoModel.register(HiggsAudioConfig, HiggsAudioModel)