GGUF是一种量化方法，是LLM库的C++复制品，支持多种LLM，如LLaMA系列和Falcon等。它允许用户在 CPU 上运行 LLM，同时将其部分层次转移到 GPU 上以加速运行。尽管使用 CPU 通常比使用 GPU 进行推理要慢，但对于在 CPU 或 Apple 设备上运行模型的人来说，这是一种非常好的格式。特别是我们看到出现了更小、更强大的模型，如 Mistral 7B，GGUF 格式可能会成为一种常见的格式它提供了从2到8位精度的不同级别的量化。我们可以获取原始的LLaMA模型，将其转换为GGUF格式，最后将GGUF格式量化为较低的精度。
Image & Text ：我们利用来自LAION-2B的图像-文本对（Schuhmann 等人，2022）、LAION-COCO（lai，2022 b）、LAION-Aesthetics（lai，2022 a）和JouneyDB（Pan 等人，2023年）。LAION-2B提供了与来自网络的嘈杂替代文本配对的图像，而LAION-COCO代表了其中的600 M子集，由BLIP标题。我们通过过滤文本质量、图像纵横比和剪辑得分等来细化这些数据集，产生了3亿对的高质量语料库。为了提高整体图像生成的保真度，我们用高质量的LAION-Aesthetics子集和Midjourney的合成数据集ColonneyDB补充了我们的数据。
Speech & Text ：我们收集了几个大规模的英语自动语音识别（ASR）数据集，Gigaspeech (Chen et al., 2021), Common Voice (Ardila et al., 2020), and Multilingual LibriSpeech(MLS) (Pratap et al., 2020). 共同构成了57，000小时的语音文本对语料库，涵盖了各种各样的说话人，领域和录音环境。
###################################
# Sample a speaker from Gaussian.
rand_spk = chat.sample_random_speaker()
print(rand_spk) # save it for later timbre recovery
params_infer_code = ChatTTS.Chat.InferCodeParams(
spk_emb = rand_spk, # add sampled speaker
temperature = .3, # using custom temperature
top_P = 0.7, # top P decode
top_K = 20, # top K decode
)
###################################
# For sentence level manual control.
# use oral_(0-9), laugh_(0-2), break_(0-7)
# to generate special token in text to synthesize.
params_refine_text = ChatTTS.Chat.RefineTextParams(
prompt='[oral_2][laugh_0][break_6]',
)
wavs = chat.infer(
texts,
params_refine_text=params_refine_text,
params_infer_code=params_infer_code,
)
###################################
# For word level manual control.
text = 'What is [uv_break]your favorite english food?[laugh][lbreak]'
wavs = chat.infer(text, skip_refine_text=True, params_refine_text=params_refine_text, params_infer_code=params_infer_code)
torchaudio.save("output2.wav", torch.from_numpy(wavs[0]), 24000)
Model Implementations for Inference（MII）是一个开源的存储库，通过减轻应用复杂系统优化技术的需求，使所有数据科学家都可以访问低延迟和高吞吐量的推理。开箱即用，MII为数千种广泛使用的DL模型提供支持，使用DeepSpeed-Inference进行优化，可以使用几行代码进行部署，同时与其香草开源版本相比，实现了显着的延迟减少。
图1：所提出的CosyVoice模型的概述。(a)演示了 𝒮3 标记器，其中虚线模块仅在训练阶段使用。(b)是CosyVoice的示意图，由文本到令牌LLM和令牌到语音流匹配模型组成。 S、E和T表示“start of sequence”、“end of sequence”和“turn of speech”标记。虚线表示推理阶段的自回归解码。(c)提供了我们的流匹配模型的放大视图，该模型以概率密度路径上的时间步长 t处的说话者嵌入 𝐯、语义标记 μ、掩蔽语音特征 X~和中间状态 Xt为条件。图2：（a）零触发上下文学习和（B）跨语言语音克隆的序列构建。LID表示目标语言标识符。
S and E denote the start and end of sequence, respectively.T is “turn of speech” tokens. 𝐯 is a speaker embedding vector extracted from the speech X with a pre-trained voice-print model2. The text encodings Y¯={𝐲¯u}u∈[1:U] is obtained by passing the text through a Byte Pair Encoded (BPE) tokenizer and text encoder:
System:
You are a translator proficient in English and {language}. Your task is to translate the following English text into {language}, focusing on a natural and fluent result that avoids “translationese.” Please consider these points:
1. Keep proper nouns, brands, and geographical names in English.
2. Retain technical terms or jargon in English, but feel free to explain in {language} if necessary.
3. Use {language} idiomatic expressions for English idioms or proverbs to ensure cultural relevance.
4. Ensure quotes or direct speech sound natural in {language}, maintaining the original’s tone.
5. For acronyms, provide the full form in {language} with the English acronym in parentheses.
User:
Text for translation: {text}
Assistant:
{translation results}
24年6月，LeCun&谢赛宁团队发布Cambrian-1，关注以视觉为中心的多模态LLM，开源了8B/13B/34B的模型。当前多模态LLM仍存在较大的视觉缺陷，需要增强视觉表征以更好地和语言模态交互，赋予模型在真实场景更强的感知定位能力。这项研究的一大意义在于影响多模态LLM的工作开始重视视觉表征质量的提升，而非一直scale up LLM。
在语音识别中，我们的数据集是音频文件和其对应的文本，不幸的是，音频文件和文本很难在单词的单位上对齐。除了语言识别，在OCR，机器翻译中，都存在类似的Sequence to Sequence结构，同样也需要在预处理操作时进行对齐，但是这种对齐有时候是非常困难的。如果不使用对齐而直接训练模型时，由于人的语速的不同，或者字符间距离的不同，导致模型很难收敛。
当我们主要关注文本和语音模态时，GPT-4o其实就是一个语音语言模型(speech language model, SLM)。该SLM同时具备语音理解能力和语音合成能力，输入端和输出端均支持文本和语音的混合多模态。那么，这一SLM应该如何实现呢？在大语言模型(large language model, LLM)滥觞的今日，不难想到这样一种方法：将连续的语音数据离散化成如同单词(或者称token，词元)一样的表示，并入到LLM的词表中，再走一遍训练LLM的老路。
audio & text tokenizer的实现应该是语音离散化部分所用的技术，例如SoundStream、Encodec、SpeechTokenizer，或者是MEL+VQ最后配合声码器来解码；参考zero-shot TTS、AudioLM/AudioPaLM、SpeechGPT-Gen等工作的结果，LLM中语音token的解码应该是要走层次化或者多步的方法，先解码语义特征，再解码声学特征，或者是先解码MEL，再加一个HIFIGAN这样的声码器。另外，如果做audio/speech/music这样的通用声合成的话，可能也能通过prompt来控制。AudioLDM2虽然做了这方面的工作，但audio/music和speech的参数其实是不一样的，说到底还不是同一个模型。
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