As two of the five traditional human senses (sight, hearing, taste, smell, and touch), vision and sound are basic sources through which humans understand the world. Often correlated during natural events, these two modalities combine to jointly affect human perception. In this paper, we pose the task of generating sound given visual input. Such capabilities could help enable applications in virtual reality (generating sound for virtual scenes automatically) or provide additional accessibility to images or videos for people with visual impairments. As a first step in this direction, we apply learning-based methods to generate raw waveform samples given input video frames. We evaluate our models on a dataset of videos containing a variety of sounds (such as ambient sounds and sounds from people animals). Our experiments show that the generated sounds are fairly realistic and have good temporal synchronization with the visual inputs.
Speech synthesis: The task of speech synthesis is to generate human speech based on input text. Text to speech (TTS) has been studied for a long time from traditional approaches @cite_24 @cite_22 @cite_21 @cite_1 @cite_13 to deep learning based approaches @cite_15 @cite_23 @cite_3 , among which, WaveNet @cite_15 has attracted much attention due to the improved generation quality. WaveNet presents a convolutional neural networks with dilation structure to predict new audio digits based on previously generated digits. SampleRNN @cite_23 also achieves appealing results in TTS, proposing a hierarchical recurrent neural network (RNN) to recursively generate raw waveform samples temporally. Its hierarchical RNN structure shows the potential to handle long sequence generation. A follow-up work @cite_3 demonstrates a novel Reader-Vocoder model and uses SampleRNN @cite_23 as the vocoder to generate raw speech signals.
@cite_22 @cite_21 @cite_1 @cite_3 @cite_24 @cite_23 @cite_15 @cite_13
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We describe a neural network-based system for text-to-speech (TTS) synthesis that is able to generate speech audio in the voice of many different speakers, including those unseen during training. Our system consists of three independently trained components: (1) a speaker encoder network, trained on a speaker verification task using an independent dataset of noisy speech from thousands of speakers without transcripts, to generate a fixed-dimensional embedding vector from seconds of reference speech from a target speaker; (2) a sequence-to-sequence synthesis network based on Tacotron 2, which generates a mel spectrogram from text, conditioned on the speaker embedding; (3) an auto-regressive WaveNet-based vocoder that converts the mel spectrogram into a sequence of time domain waveform samples. We demonstrate that the proposed model is able to transfer the knowledge of speaker variability learned by the discriminatively-trained speaker encoder to the new task, and is able to synthesize natural speech from speakers that were not seen during training. We quantify the importance of training the speaker encoder on a large and diverse speaker set in order to obtain the best generalization performance. Finally, we show that randomly sampled speaker embeddings can be used to synthesize speech in the voice of novel speakers dissimilar from those used in training, indicating that the model has learned a high quality speaker representation. Generating expressive, naturally sounding, speech from text using a speech synthesis (TTS) system is a highly challenging problem. However for tasks such as audiobooks it is essential if their use is to become widespread. Generating expressive speech from text can be divided into two parts: predicting expressive information from text; and synthesizing the speech with a particular expression. Traditionally these components have been studied separately. This paper proposes an integrated approach, where the training data and representation of expressive synthesis is shared across the two components. There are several advantages to this scheme including: robust handling of automatically generated expressive labels; support for a continuous representation of expressions; and joint training of the expression predictor and speech synthesizer. Synthesis experiments indicated that the proposed approach produced far more expressive speech than both a neutral TTS and one where the expression was randomly selected. The experimental results also show the advantage of a continuous expressive synthesis space over a discrete space. HMM-TTS synthesis is a popular approach toward flexible, low-footprint, data driven systems that produce highly intelligible speech. In spite of these strengths, speech generated by these systems exhibit some degradation in quality, attributable to an inadequacy in modeling the excitation signal that drives the parametric models of the vocal tract. This paper proposes a novel method for modeling the excitation as a low-dimensional set of coefficients, based on a non-linear map learned through an autoencoder. Through analysis-and-resynthesis experiments, and a formal listening test, we show that this model produces speech of higher perceptual quality compared to conventional pulse-excited speech signals at the p ≪ 0.01 significance level. This paper describes a novel text-to-speech (TTS) technique based on deep convolutional neural networks (CNN), without use of any recurrent units. Recurrent neural networks (RNN) have become a standard technique to model sequential data recently, and this technique has been used in some cutting-edge neural TTS techniques. However, training RNN components often requires a very powerful computer, or a very long time, typically several days or weeks. Recent other studies, on the other hand, have shown that CNN-based sequence synthesis can be much faster than RNN-based techniques, because of high parallelizability. The objective of this paper is to show that an alternative neural TTS based only on CNN alleviate these economic costs of training. In our experiment, the proposed Deep Convolutional TTS was sufficiently trained overnight (15 hours), using an ordinary gaming PC equipped with two GPUs, while the quality of the synthesized speech was almost acceptable.