40 research outputs found
Evaluation of putative reference genes for gene expression normalization in soybean by quantitative real-time RT-PCR
<p>Abstract</p> <p>Background</p> <p>Real-time quantitative reverse transcription PCR (RT-qPCR) data needs to be normalized for its proper interpretation. Housekeeping genes are routinely employed for this purpose, but their expression level cannot be assumed to remain constant under all possible experimental conditions. Thus, a systematic validation of reference genes is required to ensure proper normalization. For soybean, only a small number of validated reference genes are available to date.</p> <p>Results</p> <p>A systematic comparison of 14 potential reference genes for soybean is presented. These included seven commonly used (<it>ACT2, ACT11, TUB4, TUA5, CYP, UBQ10, EF1b</it>) and seven new candidates (<it>SKIP16, MTP, PEPKR1, HDC, TIP41, UKN1, UKN2</it>). Expression stability was examined by RT-qPCR across 116 biological samples, representing tissues at various developmental stages, varied photoperiodic treatments, and a range of soybean cultivars. Expression of all 14 genes was variable to some extent, but that of <it>SKIP16, UKN1 </it>and <it>UKN2 </it>was overall the most stable. A combination of <it>ACT11, UKN1 </it>and <it>UKN2 </it>would be appropriate as a reference panel for normalizing gene expression data among different tissues, whereas the combination SKIP16, UKN1 and MTP was most suitable for developmental stages. <it>ACT11, TUA5 </it>and <it>TIP41 </it>were the most stably expressed when the photoperiod was altered, and <it>TIP41, UKN1 </it>and <it>UKN2 </it>when the light quality was changed. For six different cultivars in long day (LD) and short day (SD), their expression stability did not vary significantly with <it>ACT11, UKN2 </it>and <it>TUB4 </it>being the most stable genes. The relative gene expression level of <it>GmFTL3</it>, an ortholog of Arabidopsis <it>FT </it>(<it>FLOWERING LOCUS T</it>) was detected to validate the reference genes selected in this study.</p> <p>Conclusion</p> <p>None of the candidate reference genes was uniformly expressed across all experimental conditions, and the most suitable reference genes are conditional-, tissue-specific-, developmental-, and cultivar-dependent. Most of the new reference genes performed better than the conventional housekeeping genes. These results should guide the selection of reference genes for gene expression studies in soybean.</p
System Fingerprint Recognition for Deepfake Audio: An Initial Dataset and Investigation
The malicious use of deep speech synthesis models may pose significant threat
to society. Therefore, many studies have emerged to detect the so-called
``deepfake audio". However, these studies focus on the binary detection of real
audio and fake audio. For some realistic application scenarios, it is needed to
know what tool or model generated the deepfake audio. This raises a question:
Can we recognize the system fingerprints of deepfake audio? Therefore, in this
paper, we propose a deepfake audio dataset for system fingerprint recognition
(SFR) and conduct an initial investigation. We collected the dataset from five
speech synthesis systems using the latest state-of-the-art deep learning
technologies, including both clean and compressed sets. In addition, to
facilitate the further development of system fingerprint recognition methods,
we give researchers some benchmarks that can be compared, and research
findings. The dataset will be publicly available.Comment: 12 pages, 3 figures. arXiv admin note: text overlap with
arXiv:2208.0964
Low-rank Adaptation Method for Wav2vec2-based Fake Audio Detection
Self-supervised speech models are a rapidly developing research topic in fake
audio detection. Many pre-trained models can serve as feature extractors,
learning richer and higher-level speech features. However,when fine-tuning
pre-trained models, there is often a challenge of excessively long training
times and high memory consumption, and complete fine-tuning is also very
expensive. To alleviate this problem, we apply low-rank adaptation(LoRA) to the
wav2vec2 model, freezing the pre-trained model weights and injecting a
trainable rank-decomposition matrix into each layer of the transformer
architecture, greatly reducing the number of trainable parameters for
downstream tasks. Compared with fine-tuning with Adam on the wav2vec2 model
containing 317M training parameters, LoRA achieved similar performance by
reducing the number of trainable parameters by 198 times.Comment: 6page
Learning to Behave Like Clean Speech: Dual-Branch Knowledge Distillation for Noise-Robust Fake Audio Detection
Most research in fake audio detection (FAD) focuses on improving performance
on standard noise-free datasets. However, in actual situations, there is
usually noise interference, which will cause significant performance
degradation in FAD systems. To improve the noise robustness, we propose a
dual-branch knowledge distillation fake audio detection (DKDFAD) method.
Specifically, a parallel data flow of the clean teacher branch and the noisy
student branch is designed, and interactive fusion and response-based
teacher-student paradigms are proposed to guide the training of noisy data from
the data distribution and decision-making perspectives. In the noise branch,
speech enhancement is first introduced for denoising, which reduces the
interference of strong noise. The proposed interactive fusion combines
denoising features and noise features to reduce the impact of speech distortion
and seek consistency with the data distribution of clean branch. The
teacher-student paradigm maps the student's decision space to the teacher's
decision space, making noisy speech behave as clean. In addition, a joint
training method is used to optimize the two branches to achieve global
optimality. Experimental results based on multiple datasets show that the
proposed method performs well in noisy environments and maintains performance
in cross-dataset experiments
Learning Speech Representation From Contrastive Token-Acoustic Pretraining
For fine-grained generation and recognition tasks such as
minimally-supervised text-to-speech (TTS), voice conversion (VC), and automatic
speech recognition (ASR), the intermediate representations extracted from
speech should serve as a "bridge" between text and acoustic information,
containing information from both modalities. The semantic content is
emphasized, while the paralinguistic information such as speaker identity and
acoustic details should be de-emphasized. However, existing methods for
extracting fine-grained intermediate representations from speech suffer from
issues of excessive redundancy and dimension explosion. Contrastive learning is
a good method for modeling intermediate representations from two modalities.
However, existing contrastive learning methods in the audio field focus on
extracting global descriptive information for downstream audio classification
tasks, making them unsuitable for TTS, VC, and ASR tasks. To address these
issues, we propose a method named "Contrastive Token-Acoustic Pretraining
(CTAP)", which uses two encoders to bring phoneme and speech into a joint
multimodal space, learning how to connect phoneme and speech at the frame
level. The CTAP model is trained on 210k speech and phoneme text pairs,
achieving minimally-supervised TTS, VC, and ASR. The proposed CTAP method
offers a promising solution for fine-grained generation and recognition
downstream tasks in speech processing
CFAD: A Chinese Dataset for Fake Audio Detection
Fake audio detection is a growing concern and some relevant datasets have
been designed for research. However, there is no standard public Chinese
dataset under complex conditions.In this paper, we aim to fill in the gap and
design a Chinese fake audio detection dataset (CFAD) for studying more
generalized detection methods. Twelve mainstream speech-generation techniques
are used to generate fake audio. To simulate the real-life scenarios, three
noise datasets are selected for noise adding at five different signal-to-noise
ratios, and six codecs are considered for audio transcoding (format
conversion). CFAD dataset can be used not only for fake audio detection but
also for detecting the algorithms of fake utterances for audio forensics.
Baseline results are presented with analysis. The results that show fake audio
detection methods with generalization remain challenging. The CFAD dataset is
publicly available at: https://zenodo.org/record/8122764.Comment: FAD renamed as CFA
Minimally-Supervised Speech Synthesis with Conditional Diffusion Model and Language Model: A Comparative Study of Semantic Coding
Recently, there has been a growing interest in text-to-speech (TTS) methods
that can be trained with minimal supervision by combining two types of discrete
speech representations and using two sequence-to-sequence tasks to decouple
TTS. To address the challenges associated with high dimensionality and waveform
distortion in discrete representations, we propose Diff-LM-Speech, which models
semantic embeddings into mel-spectrogram based on diffusion models and
introduces a prompt encoder structure based on variational autoencoders and
prosody bottlenecks to improve prompt representation capabilities.
Autoregressive language models often suffer from missing and repeated words,
while non-autoregressive frameworks face expression averaging problems due to
duration prediction models. To address these issues, we propose
Tetra-Diff-Speech, which designs a duration diffusion model to achieve diverse
prosodic expressions. While we expect the information content of semantic
coding to be between that of text and acoustic coding, existing models extract
semantic coding with a lot of redundant information and dimensionality
explosion. To verify that semantic coding is not necessary, we propose
Tri-Diff-Speech. Experimental results show that our proposed methods outperform
baseline methods. We provide a website with audio samples
Chinese Open Instruction Generalist: A Preliminary Release
Instruction tuning is widely recognized as a key technique for building
generalist language models, which has attracted the attention of researchers
and the public with the release of InstructGPT~\citep{ouyang2022training} and
ChatGPT\footnote{\url{https://chat.openai.com/}}. Despite impressive progress
in English-oriented large-scale language models (LLMs), it is still
under-explored whether English-based foundation LLMs can perform similarly on
multilingual tasks compared to English tasks with well-designed instruction
tuning and how we can construct the corpora needed for the tuning.
To remedy this gap, we propose the project as an attempt to create a Chinese
instruction dataset by various methods adapted to the intrinsic characteristics
of 4 sub-tasks. We collect around 200k Chinese instruction tuning samples,
which have been manually checked to guarantee high quality. We also summarize
the existing English and Chinese instruction corpora and briefly describe some
potential applications of the newly constructed Chinese instruction corpora.
The resulting \textbf{C}hinese \textbf{O}pen \textbf{I}nstruction
\textbf{G}eneralist (\textbf{COIG}) corpora are available in
Huggingface\footnote{\url{https://huggingface.co/datasets/BAAI/COIG}} and
Github\footnote{\url{https://github.com/FlagOpen/FlagInstruct}}, and will be
continuously updated
An Overview of Affective Speech Synthesis and Conversion in the Deep Learning Era
Speech is the fundamental mode of human communication, and its synthesis has
long been a core priority in human-computer interaction research. In recent
years, machines have managed to master the art of generating speech that is
understandable by humans. But the linguistic content of an utterance
encompasses only a part of its meaning. Affect, or expressivity, has the
capacity to turn speech into a medium capable of conveying intimate thoughts,
feelings, and emotions -- aspects that are essential for engaging and
naturalistic interpersonal communication. While the goal of imparting
expressivity to synthesised utterances has so far remained elusive, following
recent advances in text-to-speech synthesis, a paradigm shift is well under way
in the fields of affective speech synthesis and conversion as well. Deep
learning, as the technology which underlies most of the recent advances in
artificial intelligence, is spearheading these efforts. In the present
overview, we outline ongoing trends and summarise state-of-the-art approaches
in an attempt to provide a comprehensive overview of this exciting field.Comment: Submitted to the Proceedings of IEE
On the Effectiveness of Speech Self-supervised Learning for Music
Self-supervised learning (SSL) has shown promising results in various speech
and natural language processing applications. However, its efficacy in music
information retrieval (MIR) still remains largely unexplored. While previous
SSL models pre-trained on music recordings may have been mostly closed-sourced,
recent speech models such as wav2vec2.0 have shown promise in music modelling.
Nevertheless, research exploring the effectiveness of applying speech SSL
models to music recordings has been limited. We explore the music adaption of
SSL with two distinctive speech-related models, data2vec1.0 and Hubert, and
refer to them as music2vec and musicHuBERT, respectively. We train SSL
models with 95M parameters under various pre-training configurations and
systematically evaluate the MIR task performances with 13 different MIR tasks.
Our findings suggest that training with music data can generally improve
performance on MIR tasks, even when models are trained using paradigms designed
for speech. However, we identify the limitations of such existing
speech-oriented designs, especially in modelling polyphonic information. Based
on the experimental results, empirical suggestions are also given for designing
future musical SSL strategies and paradigms