prompt_front: You are given a scientific article and a question. Answer the question as concisely as you can, using a single phrase or sentence if possible. If the question cannot be answered based on the information in the article, write "unanswerable". If the question is a yes/no question, answer "yes", "no", or "unanswerable". Do not provide any explanation.

Please answer the questions based on the text provided below. How big is Augmented LibriSpeech dataset?
Context 1: Introduction
End-to-end speech-to-text translation (ST) has attracted much attention recently BIBREF2, BIBREF3, BIBREF4, BIBREF5, BIBREF6 given its simplicity against cascading automatic speech recognition (ASR) and machine translation (MT) systems  The lack of labeled data, however, has become a major blocker for bridging the performance gaps between end-to-end models and cascading systems  Several corpora have been developed in recent years  post2013improved introduced a 38-hour Spanish-English ST corpus by augmenting the transcripts of the Fisher and Callhome corpora with English translations  di-gangi-etal-2019-must created the largest ST corpus to date from TED talks but the language pairs involved are out of English only  beilharz2019librivoxdeen created a 110-hour German-English ST corpus from LibriVox audiobooks  godard-etal-2018-low created a Moboshi-French ST corpus as part of a rare language documentation effort  woldeyohannis provided an Amharic-English ST corpus in the tourism domain  boito2019mass created a multilingual ST corpus involving 8 languages from a multilingual speech corpus based on Bible readings BIBREF7  Previous work either involves language pairs out of English, very specific domains, very low resource languages or a limited set of language pairs  This limits the scope of study, including the latest explorations on end-to-end multilingual ST BIBREF8, BIBREF9  Our work is mostly similar and concurrent to iranzosnchez2019europarlst who created a multilingual ST corpus from the European Parliament proceedings  The corpus we introduce has larger speech durations and more translation tokens  It is diversified with multiple speakers per transcript/translation  Finally, we provide additional out-of-domain test sets 
In this paper, we introduce CoVoST, a multilingual ST corpus based on Common Voice BIBREF10 for 11 languages into English, diversified with over 11,000 speakers and over 60 accents  It includes a total 708 hours of French (Fr), German (De), Dutch (Nl), Russian (Ru), Spanish (Es), Italian (It), Turkish (Tr), Persian (Fa), Swedish (Sv), Mongolian (Mn) and Chinese (Zh) speeches, with French and German ones having the largest durations among existing public corpora  We also collect an additional evaluation corpus from Tatoeba for French, German, Dutch, Russian and Spanish, resulting in a total of 9 3 hours of speech  Both corpora are created at the sentence level and do not require additional alignments or segmentation  Using the official Common Voice train-development-test split, we also provide baseline models, including, to our knowledge, the first end-to-end many-to-one multilingual ST models  CoVoST is released under CC0 license and free to use  The Tatoeba evaluation samples are also available under friendly CC licenses  All the data can be acquired at https://github com/facebookresearch/covost 
Data Collection and Processing ::: Common Voice (CoVo)
Common Voice BIBREF10 is a crowdsourcing speech recognition corpus with an open CC0 license  Contributors record voice clips by reading from a bank of donated sentences  Each voice clip was validated by at least two other users  Most of the sentences are covered by multiple speakers, with potentially different genders, age groups or accents 
Raw CoVo data contains samples that passed validation as well as those that did not  To build CoVoST, we only use the former one and reuse the official train-development-test partition of the validated data  As of January 2020, the latest CoVo 2019-06-12 release includes 29 languages  CoVoST is currently built on that release and covers the following 11 languages: French, German, Dutch, Russian, Spanish, Italian, Turkish, Persian, Swedish, Mongolian and Chinese 
Validated transcripts were sent to professional translators  Note that the translators had access to the transcripts but not the corresponding voice clips since clips would not carry additional information  Since transcripts were duplicated due to multiple speakers, we deduplicated the transcripts before sending them to translators  As a result, different voice clips of the same content (transcript) will have identical translations in CoVoST for train, development and test splits 
In order to control the quality of the professional translations, we applied various sanity checks to the translations BIBREF11  1) For German-English, French-English and Russian-English translations, we computed sentence-level BLEU BIBREF12 with the NLTK BIBREF13 implementation between the human translations and the automatic translations produced by a state-of-the-art system BIBREF14 (the French-English system was a Transformer big BIBREF15 separately trained on WMT14)  We applied this method to these three language pairs only as we are confident about the quality of the corresponding systems  Translations with a score that was too low were manually inspected and sent back to the translators when needed  2) We manually inspected examples where the source transcript was identical to the translation  3) We measured the perplexity of the translations using a language model trained on a large amount of clean monolingual data BIBREF14  We manually inspected examples where the translation had a high perplexity and sent them back to translators accordingly  4) We computed the ratio of English characters in the translations  We manually inspected examples with a low ratio and sent them back to translators accordingly  5) Finally, we used VizSeq BIBREF16 to calculate similarity scores between transcripts and translations based on LASER cross-lingual sentence embeddings BIBREF17  Samples with low scores were manually inspected and sent back for translation when needed 
We also sanity check the overlaps of train, development and test sets in terms of transcripts and voice clips (via MD5 file hashing), and confirm they are totally disjoint 
Data Collection and Processing ::: Tatoeba (TT)
Tatoeba (TT) is a community built language learning corpus having sentences aligned across multiple languages with the corresponding speech partially available  Its sentences are on average shorter than those in CoVoST (see also Table TABREF2) given the original purpose of language learning  Sentences in TT are licensed under CC BY 2 0 FR and part of the speeches are available under various CC licenses 
We construct an evaluation set from TT (for French, German, Dutch, Russian and Spanish) as a complement to CoVoST development and test sets  We collect (speech, transcript, English translation) triplets for the 5 languages and do not include those whose speech has a broken URL or is not CC licensed  We further filter these samples by sentence lengths (minimum 4 words including punctuations) to reduce the portion of short sentences  This makes the resulting evaluation set closer to real-world scenarios and more challenging 
We run the same quality checks for TT as for CoVoST but we do not find poor quality translations according to our criteria  Finally, we report the overlap between CoVo transcripts and TT sentences in Table TABREF5  We found a minimal overlap, which makes the TT evaluation set a suitable additional test set when training on CoVoST 
Data Analysis ::: Basic Statistics
Basic statistics for CoVoST and TT are listed in Table TABREF2 including (unique) sentence counts, speech durations, speaker demographics (partially available) as well as vocabulary and token statistics (based on Moses-tokenized sentences by sacreMoses) on both transcripts and translations  We see that CoVoST has over 327 hours of German speeches and over 171 hours of French speeches, which, to our knowledge, corresponds to the largest corpus among existing public ST corpora (the second largest is 110 hours BIBREF18 for German and 38 hours BIBREF19 for French)  Moreover, CoVoST has a total of 18 hours of Dutch speeches, to our knowledge, contributing the first public Dutch ST resource  CoVoST also has around 27-hour Russian speeches, 37-hour Italian speeches and 67-hour Persian speeches, which is 1 8 times, 2 5 times and 13 3 times of the previous largest public one BIBREF7  Most of the sentences (transcripts) in CoVoST are covered by multiple speakers with potentially different accents, resulting in a rich diversity in the speeches  For example, there are over 1,000 speakers and over 10 accents in the French and German development / test sets  This enables good coverage of speech variations in both model training and evaluation 
Data Analysis ::: Speaker Diversity
As we can see from Table TABREF2, CoVoST is diversified with a rich set of speakers and accents  We further inspect the speaker demographics in terms of sample distributions with respect to speaker counts, accent counts and age groups, which is shown in Figure FIGREF6, FIGREF7 and FIGREF8  We observe that for 8 of the 11 languages, at least 60% of the sentences (transcripts) are covered by multiple speakers  Over 80% of the French sentences have at least 3 speakers  And for German sentences, even over 90% of them have at least 5 speakers  Similarly, we see that a large portion of sentences are spoken in multiple accents for French, German, Dutch and Spanish  Speakers of each language also spread widely across different age groups (below 20, 20s, 30s, 40s, 50s, 60s and 70s) 
Baseline Results
We provide baselines using the official train-development-test split on the following tasks: automatic speech recognition (ASR), machine translation (MT) and speech translation (ST) 
Baseline Results ::: Experimental Settings ::: Data Preprocessing
We convert raw MP3 audio files from CoVo and TT into mono-channel waveforms, and downsample them to 16,000 Hz  For transcripts and translations, we normalize the punctuation, we tokenize the text with sacreMoses and lowercase it  For transcripts, we further remove all punctuation markers except for apostrophes  We use character vocabularies on all the tasks, with 100% coverage of all the characters  Preliminary experimentation showed that character vocabularies provided more stable training than BPE  For MT, the vocabulary is created jointly on both transcripts and translations  We extract 80-channel log-mel filterbank features, computed with a 25ms window size and 10ms window shift using torchaudio  The features are normalized to 0 mean and 1 0 standard deviation  We remove samples having more than 3,000 frames or more than 256 characters for GPU memory efficiency (less than 25 samples are removed for all languages) 
Baseline Results ::: Experimental Settings ::: Model Training
Our ASR and ST models follow the architecture in berard2018end, but have 3 decoder layers like that in pino2019harnessing  For MT, we use a Transformer base architecture BIBREF15, but with 3 encoder layers, 3 decoder layers and 0 3 dropout  We use a batch size of 10,000 frames for ASR and ST, and a batch size of 4,000 tokens for MT  We train all models using Fairseq BIBREF20 for up to 200,000 updates  We use SpecAugment BIBREF21 for ASR and ST to alleviate overfitting 
Baseline Results ::: Experimental Settings ::: Inference and Evaluation
We use a beam size of 5 for all models  We use the best checkpoint by validation loss for MT, and average the last 5 checkpoints for ASR and ST  For MT and ST, we report case-insensitive tokenized BLEU BIBREF22 using sacreBLEU BIBREF23  For ASR, we report word error rate (WER) and character error rate (CER) using VizSeq 
Baseline Results ::: Automatic Speech Recognition (ASR)
For simplicity, we use the same model architecture for ASR and ST, although we do not leverage ASR models to pretrain ST model encoders later  Table TABREF18 shows the word error rate (WER) and character error rate (CER) for ASR models  We see that French and German perform the best given they are the two highest resource languages in CoVoST  The other languages are relatively low resource (especially Turkish and Swedish) and the ASR models are having difficulties to learn from this data 
Baseline Results ::: Machine Translation (MT)
MT models take transcripts (without punctuation) as inputs and outputs translations (with punctuation)  For simplicity, we do not change the text preprocessing methods for MT to correct this mismatch  Moreover, this mismatch also exists in cascading ST systems, where MT model inputs are the outputs of an ASR model  Table TABREF20 shows the BLEU scores of MT models  We notice that the results are consistent with what we see from ASR models  For example thanks to abundant training data, French has a decent BLEU score of 29 8/25 4  German doesn't perform well, because of less richness of content (transcripts)  The other languages are low resource in CoVoST and it is difficult to train decent models without additional data or pre-training techniques 
Baseline Results ::: Speech Translation (ST)
CoVoST is a many-to-one multilingual ST corpus  While end-to-end one-to-many and many-to-many multilingual ST models have been explored very recently BIBREF8, BIBREF9, many-to-one multilingual models, to our knowledge, have not  We hence use CoVoST to examine this setting  Table TABREF22 and TABREF23 show the BLEU scores for both bilingual and multilingual end-to-end ST models trained on CoVoST  We observe that combining speeches from multiple languages is consistently bringing gains to low-resource languages (all besides French and German)  This includes combinations of distant languages, such as Ru+Fr, Tr+Fr and Zh+Fr  Moreover, some combinations do bring gains to high-resource language (French) as well: Es+Fr, Tr+Fr and Mn+Fr  We simply provide the most basic many-to-one multilingual baselines here, and leave the full exploration of the best configurations to future work  Finally, we note that for some language pairs, absolute BLEU numbers are relatively low as we restrict model training to the supervised data  We encourage the community to improve upon those baselines, for example by leveraging semi-supervised training 
Baseline Results ::: Multi-Speaker Evaluation
In CoVoST, large portion of transcripts are covered by multiple speakers with different genders, accents and age groups  Besides the standard corpus-level BLEU scores, we also want to evaluate model output variance on the same content (transcript) but different speakers  We hence propose to group samples (and their sentence BLEU scores) by transcript, and then calculate average per-group mean and average coefficient of variation defined as follows:
and
where $G$ is the set of sentence BLEU scores grouped by transcript and $G^{\prime } = \lbrace g | g\in G, |g|>1, \textrm {Mean}(g) > 0 \rbrace $ 
$\textrm {BLEU}_{MS}$ provides a normalized quality score as oppose to corpus-level BLEU or unnormalized average of sentence BLEU  And $\textrm {CoefVar}_{MS}$ is a standardized measure of model stability against different speakers (the lower the better)  Table TABREF24 shows the $\textrm {BLEU}_{MS}$ and $\textrm {CoefVar}_{MS}$ of our ST models on CoVoST test set  We see that German and Persian have the worst $\textrm {CoefVar}_{MS}$ (least stable) given their rich speaker diversity in the test set and relatively small train set (see also Figure FIGREF6 and Table TABREF2)  Dutch also has poor $\textrm {CoefVar}_{MS}$ because of the lack of training data  Multilingual models are consistantly more stable on low-resource languages  Ru+Fr, Tr+Fr, Fa+Fr and Zh+Fr even have better $\textrm {CoefVar}_{MS}$ than all individual languages 
Conclusion
We introduce a multilingual speech-to-text translation corpus, CoVoST, for 11 languages into English, diversified with over 11,000 speakers and over 60 accents  We also provide baseline results, including, to our knowledge, the first end-to-end many-to-one multilingual model for spoken language translation  CoVoST is free to use with a CC0 license, and the additional Tatoeba evaluation samples are also CC-licensed 
Context 2: Introduction
In the kitchen, we increasingly rely on instructions from cooking websites: recipes  A cook with a predilection for Asian cuisine may wish to prepare chicken curry, but may not know all necessary ingredients apart from a few basics  These users with limited knowledge cannot rely on existing recipe generation approaches that focus on creating coherent recipes given all ingredients and a recipe name BIBREF0  Such models do not address issues of personal preference (e g  culinary tastes, garnish choices) and incomplete recipe details  We propose to approach both problems via personalized generation of plausible, user-specific recipes using user preferences extracted from previously consumed recipes 
Our work combines two important tasks from natural language processing and recommender systems: data-to-text generation BIBREF1 and personalized recommendation BIBREF2  Our model takes as user input the name of a specific dish, a few key ingredients, and a calorie level  We pass these loose input specifications to an encoder-decoder framework and attend on user profiles—learned latent representations of recipes previously consumed by a user—to generate a recipe personalized to the user's tastes  We fuse these `user-aware' representations with decoder output in an attention fusion layer to jointly determine text generation  Quantitative (perplexity, user-ranking) and qualitative analysis on user-aware model outputs confirm that personalization indeed assists in generating plausible recipes from incomplete ingredients 
While personalized text generation has seen success in conveying user writing styles in the product review BIBREF3, BIBREF4 and dialogue BIBREF5 spaces, we are the first to consider it for the problem of recipe generation, where output quality is heavily dependent on the content of the instructions—such as ingredients and cooking techniques 
To summarize, our main contributions are as follows:
We explore a new task of generating plausible and personalized recipes from incomplete input specifications by leveraging historical user preferences;
We release a new dataset of 180K+ recipes and 700K+ user reviews for this task;
We introduce new evaluation strategies for generation quality in instructional texts, centering on quantitative measures of coherence  We also show qualitatively and quantitatively that personalized models generate high-quality and specific recipes that align with historical user preferences 
Related Work
Large-scale transformer-based language models have shown surprising expressivity and fluency in creative and conditional long-text generation BIBREF6, BIBREF7  Recent works have proposed hierarchical methods that condition on narrative frameworks to generate internally consistent long texts BIBREF8, BIBREF9, BIBREF10  Here, we generate procedurally structured recipes instead of free-form narratives 
Recipe generation belongs to the field of data-to-text natural language generation BIBREF1, which sees other applications in automated journalism BIBREF11, question-answering BIBREF12, and abstractive summarization BIBREF13, among others  BIBREF14, BIBREF15 model recipes as a structured collection of ingredient entities acted upon by cooking actions  BIBREF0 imposes a `checklist' attention constraint emphasizing hitherto unused ingredients during generation  BIBREF16 attend over explicit ingredient references in the prior recipe step  Similar hierarchical approaches that infer a full ingredient list to constrain generation will not help personalize recipes, and would be infeasible in our setting due to the potentially unconstrained number of ingredients (from a space of 10K+) in a recipe  We instead learn historical preferences to guide full recipe generation 
A recent line of work has explored user- and item-dependent aspect-aware review generation BIBREF3, BIBREF4  This work is related to ours in that it combines contextual language generation with personalization  Here, we attend over historical user preferences from previously consumed recipes to generate recipe content, rather than writing styles 
Approach
Our model's input specification consists of: the recipe name as a sequence of tokens, a partial list of ingredients, and a caloric level (high, medium, low)  It outputs the recipe instructions as a token sequence: $\mathcal {W}_r=\lbrace w_{r,0}, \dots , w_{r,T}\rbrace $ for a recipe $r$ of length $T$  To personalize output, we use historical recipe interactions of a user $u \in \mathcal {U}$ 
Encoder: Our encoder has three embedding layers: vocabulary embedding $\mathcal {V}$, ingredient embedding $\mathcal {I}$, and caloric-level embedding $\mathcal {C}$  Each token in the (length $L_n$) recipe name is embedded via $\mathcal {V}$; the embedded token sequence is passed to a two-layered bidirectional GRU (BiGRU) BIBREF17, which outputs hidden states for names $\lbrace \mathbf {n}_{\text{enc},j} \in \mathbb {R}^{2d_h}\rbrace $, with hidden size $d_h$  Similarly each of the $L_i$ input ingredients is embedded via $\mathcal {I}$, and the embedded ingredient sequence is passed to another two-layered BiGRU to output ingredient hidden states as $\lbrace \mathbf {i}_{\text{enc},j} \in \mathbb {R}^{2d_h}\rbrace $  The caloric level is embedded via $\mathcal {C}$ and passed through a projection layer with weights $W_c$ to generate calorie hidden representation $\mathbf {c}_{\text{enc}} \in \mathbb {R}^{2d_h}$ 
Ingredient Attention: We apply attention BIBREF18 over the encoded ingredients to use encoder outputs at each decoding time step  We define an attention-score function $\alpha $ with key $K$ and query $Q$:
with trainable weights $W_{\alpha }$, bias $\mathbf {b}_{\alpha }$, and normalization term $Z$  At decoding time $t$, we calculate the ingredient context $\mathbf {a}_{t}^{i} \in \mathbb {R}^{d_h}$ as:
Decoder: The decoder is a two-layer GRU with hidden state $h_t$ conditioned on previous hidden state $h_{t-1}$ and input token $w_{r, t}$ from the original recipe text  We project the concatenated encoder outputs as the initial decoder hidden state:
To bias generation toward user preferences, we attend over a user's previously reviewed recipes to jointly determine the final output token distribution  We consider two different schemes to model preferences from user histories: (1) recipe interactions, and (2) techniques seen therein (defined in data)  BIBREF19, BIBREF20, BIBREF21 explore similar schemes for personalized recommendation 
Prior Recipe Attention: We obtain the set of prior recipes for a user $u$: $R^+_u$, where each recipe can be represented by an embedding from a recipe embedding layer $\mathcal {R}$ or an average of the name tokens embedded by $\mathcal {V}$  We attend over the $k$-most recent prior recipes, $R^{k+}_u$, to account for temporal drift of user preferences BIBREF22  These embeddings are used in the `Prior Recipe' and `Prior Name' models, respectively 
Given a recipe representation $\mathbf {r} \in \mathbb {R}^{d_r}$ (where $d_r$ is recipe- or vocabulary-embedding size depending on the recipe representation) the prior recipe attention context $\mathbf {a}_{t}^{r_u}$ is calculated as
Prior Technique Attention: We calculate prior technique preference (used in the `Prior Tech` model) by normalizing co-occurrence between users and techniques seen in $R^+_u$, to obtain a preference vector $\rho _{u}$  Each technique $x$ is embedded via a technique embedding layer $\mathcal {X}$ to $\mathbf {x}\in \mathbb {R}^{d_x}$  Prior technique attention is calculated as
where, inspired by copy mechanisms BIBREF23, BIBREF24, we add $\rho _{u,x}$ for technique $x$ to emphasize the attention by the user's prior technique preference 
Attention Fusion Layer: We fuse all contexts calculated at time $t$, concatenating them with decoder GRU output and previous token embedding:
We then calculate the token probability:
and maximize the log-likelihood of the generated sequence conditioned on input specifications and user preferences  fig:ex shows a case where the Prior Name model attends strongly on previously consumed savory recipes to suggest the usage of an additional ingredient (`cilantro') 
Recipe Dataset: Food com
We collect a novel dataset of 230K+ recipe texts and 1M+ user interactions (reviews) over 18 years (2000-2018) from Food com  Here, we restrict to recipes with at least 3 steps, and at least 4 and no more than 20 ingredients  We discard users with fewer than 4 reviews, giving 180K+ recipes and 700K+ reviews, with splits as in tab:recipeixnstats 
Our model must learn to generate from a diverse recipe space: in our training data, the average recipe length is 117 tokens with a maximum of 256  There are 13K unique ingredients across all recipes  Rare words dominate the vocabulary: 95% of words appear $<$100 times, accounting for only 1 65% of all word usage  As such, we perform Byte-Pair Encoding (BPE) tokenization BIBREF25, BIBREF26, giving a training vocabulary of 15K tokens across 19M total mentions  User profiles are similarly diverse: 50% of users have consumed $\le $6 recipes, while 10% of users have consumed $>$45 recipes 
We order reviews by timestamp, keeping the most recent review for each user as the test set, the second most recent for validation, and the remainder for training (sequential leave-one-out evaluation BIBREF27)  We evaluate only on recipes not in the training set 
We manually construct a list of 58 cooking techniques from 384 cooking actions collected by BIBREF15; the most common techniques (bake, combine, pour, boil) account for 36 5% of technique mentions  We approximate technique adherence via string match between the recipe text and technique list 
Experiments and Results
For training and evaluation, we provide our model with the first 3-5 ingredients listed in each recipe  We decode recipe text via top-$k$ sampling BIBREF7, finding $k=3$ to produce satisfactory results  We use a hidden size $d_h=256$ for both the encoder and decoder  Embedding dimensions for vocabulary, ingredient, recipe, techniques, and caloric level are 300, 10, 50, 50, and 5 (respectively)  For prior recipe attention, we set $k=20$, the 80th %-ile for the number of user interactions  We use the Adam optimizer BIBREF28 with a learning rate of $10^{-3}$, annealed with a decay rate of 0 9 BIBREF29  We also use teacher-forcing BIBREF30 in all training epochs
Context 3: Introduction
Recently, deep learning algorithms have successfully addressed problems in various fields, such as image classification, machine translation, speech recognition, text-to-speech generation and other machine learning related areas BIBREF0 , BIBREF1 , BIBREF2   Similarly, substantial improvements in performance have been obtained when deep learning algorithms have been applied to statistical speech processing BIBREF3   These fundamental improvements have led researchers to investigate additional topics related to human nature, which have long been objects of study  One such topic involves understanding human emotions and reflecting it through machine intelligence, such as emotional dialogue models BIBREF4 , BIBREF5  
In developing emotionally aware intelligence, the very first step is building robust emotion classifiers that display good performance regardless of the application; this outcome is considered to be one of the fundamental research goals in affective computing BIBREF6   In particular, the speech emotion recognition task is one of the most important problems in the field of paralinguistics  This field has recently broadened its applications, as it is a crucial factor in optimal human-computer interactions, including dialog systems  The goal of speech emotion recognition is to predict the emotional content of speech and to classify speech according to one of several labels (i e , happy, sad, neutral, and angry)  Various types of deep learning methods have been applied to increase the performance of emotion classifiers; however, this task is still considered to be challenging for several reasons  First, insufficient data for training complex neural network-based models are available, due to the costs associated with human involvement  Second, the characteristics of emotions must be learned from low-level speech signals  Feature-based models display limited skills when applied to this problem 
To overcome these limitations, we propose a model that uses high-level text transcription, as well as low-level audio signals, to utilize the information contained within low-resource datasets to a greater degree  Given recent improvements in automatic speech recognition (ASR) technology BIBREF7 , BIBREF2 , BIBREF8 , BIBREF9 , speech transcription can be carried out using audio signals with considerable skill  The emotional content of speech is clearly indicated by the emotion words contained in a sentence BIBREF10 , such as “lovely” and “awesome,” which carry strong emotions compared to generic (non-emotion) words, such as “person” and “day ” Thus, we hypothesize that the speech emotion recognition model will be benefit from the incorporation of high-level textual input 
In this paper, we propose a novel deep dual recurrent encoder model that simultaneously utilizes audio and text data in recognizing emotions from speech  Extensive experiments are conducted to investigate the efficacy and properties of the proposed model  Our proposed model outperforms previous state-of-the-art methods by 68 8% to 71 8% when applied to the IEMOCAP dataset, which is one of the most well-studied datasets  Based on an error analysis of the models, we show that our proposed model accurately identifies emotion classes  Moreover, the neutral class misclassification bias frequently exhibited by previous models, which focus on audio features, is less pronounced in our model 
Related work
Classical machine learning algorithms, such as hidden Markov models (HMMs), support vector machines (SVMs), and decision tree-based methods, have been employed in speech emotion recognition problems BIBREF11 , BIBREF12 , BIBREF13   Recently, researchers have proposed various neural network-based architectures to improve the performance of speech emotion recognition  An initial study utilized deep neural networks (DNNs) to extract high-level features from raw audio data and demonstrated its effectiveness in speech emotion recognition BIBREF14   With the advancement of deep learning methods, more complex neural-based architectures have been proposed  Convolutional neural network (CNN)-based models have been trained on information derived from raw audio signals using spectrograms or audio features such as Mel-frequency cepstral coefficients (MFCCs) and low-level descriptors (LLDs) BIBREF15 , BIBREF16 , BIBREF17   These neural network-based models are combined to produce higher-complexity models BIBREF18 , BIBREF19 , and these models achieved the best-recorded performance when applied to the IEMOCAP dataset 
Another line of research has focused on adopting variant machine learning techniques combined with neural network-based models  One researcher utilized the multiobject learning approach and used gender and naturalness as auxiliary tasks so that the neural network-based model learned more features from a given dataset BIBREF20   Another researcher investigated transfer learning methods, leveraging external data from related domains BIBREF21  
As emotional dialogue is composed of sound and spoken content, researchers have also investigated the combination of acoustic features and language information, built belief network-based methods of identifying emotional key phrases, and assessed the emotional salience of verbal cues from both phoneme sequences and words BIBREF22 , BIBREF23   However, none of these studies have utilized information from speech signals and text sequences simultaneously in an end-to-end learning neural network-based model to classify emotions 
Model
This section describes the methodologies that are applied to the speech emotion recognition task  We start by introducing the recurrent encoder model for the audio and text modalities individually  We then propose a multimodal approach that encodes both audio and textual information simultaneously via a dual recurrent encoder 
Audio Recurrent Encoder (ARE)
Motivated by the architecture used in BIBREF24 , BIBREF25 , we build an audio recurrent encoder (ARE) to predict the class of a given audio signal  Once MFCC features have been extracted from an audio signal, a subset of the sequential features is fed into the RNN (i e , gated recurrent units (GRUs)), which leads to the formation of the network's internal hidden state INLINEFORM0 to model the time series patterns  This internal hidden state is updated at each time step with the input data INLINEFORM1 and the hidden state of the previous time step INLINEFORM2 as follows: DISPLAYFORM0
where INLINEFORM0 is the RNN function with weight parameter INLINEFORM1 , INLINEFORM2 represents the hidden state at t- INLINEFORM3 time step, and INLINEFORM4 represents the t- INLINEFORM5 MFCC features in INLINEFORM6   After encoding the audio signal INLINEFORM7 with the RNN, the last hidden state of the RNN, INLINEFORM8 , is considered to be the representative vector that contains all of the sequential audio data  This vector is then concatenated with another prosodic feature vector, INLINEFORM9 , to generate a more informative vector representation of the signal, INLINEFORM10   The MFCC and the prosodic features are extracted from the audio signal using the openSMILE toolkit BIBREF26 , INLINEFORM11 , respectively  Finally, the emotion class is predicted by applying the softmax function to the vector INLINEFORM12   For a given audio sample INLINEFORM13 , we assume that INLINEFORM14 is the true label vector, which contains all zeros but contains a one at the correct class, and INLINEFORM15 is the predicted probability distribution from the softmax layer  The training objective then takes the following form: DISPLAYFORM0
where INLINEFORM0 is the calculated representative vector of the audio signal with dimensionality INLINEFORM1   The INLINEFORM2 and the bias INLINEFORM3 are learned model parameters  C is the total number of classes, and N is the total number of samples used in training  The upper part of Figure shows the architecture of the ARE model 
Text Recurrent Encoder (TRE)
We assume that speech transcripts can be extracted from audio signals with high accuracy, given the advancement of ASR technologies BIBREF7   We attempt to use the processed textual information as another modality in predicting the emotion class of a given signal  To use textual information, a speech transcript is tokenized and indexed into a sequence of tokens using the Natural Language Toolkit (NLTK) BIBREF27   Each token is then passed through a word-embedding layer that converts a word index to a corresponding 300-dimensional vector that contains additional contextual meaning between words  The sequence of embedded tokens is fed into a text recurrent encoder (TRE) in such a way that the audio MFCC features are encoded using the ARE represented by equation EQREF2   In this case, INLINEFORM0 is the t- INLINEFORM1 embedded token from the text input  Finally, the emotion class is predicted from the last hidden state of the text-RNN using the softmax function 
We use the same training objective as the ARE model, and the predicted probability distribution for the target class is as follows: DISPLAYFORM0
where INLINEFORM0 is last hidden state of the text-RNN, INLINEFORM1 , and the INLINEFORM2 and bias INLINEFORM3 are learned model parameters  The lower part of Figure indicates the architecture of the TRE model 
Multimodal Dual Recurrent Encoder (MDRE)
We present a novel architecture called the multimodal dual recurrent encoder (MDRE) to overcome the limitations of existing approaches  In this study, we consider multiple modalities, such as MFCC features, prosodic features and transcripts, which contain sequential audio information, statistical audio information and textual information, respectively  These types of data are the same as those used in the ARE and TRE cases  The MDRE model employs two RNNs to encode data from the audio signal and textual inputs independently  The audio-RNN encodes MFCC features from the audio signal using equation EQREF2   The last hidden state of the audio-RNN is concatenated with the prosodic features to form the final vector representation INLINEFORM0 , and this vector is then passed through a fully connected neural network layer to form the audio encoding vector A  On the other hand, the text-RNN encodes the word sequence of the transcript using equation EQREF2   The final hidden states of the text-RNN are also passed through another fully connected neural network layer to form a textual encoding vector T  Finally, the emotion class is predicted by applying the softmax function to the concatenation of the vectors A and T  We use the same training objective as the ARE model, and the predicted probability distribution for the target class is as follows: DISPLAYFORM0
where INLINEFORM0 is the feed-forward neural network with weight parameter INLINEFORM1 , and INLINEFORM2 , INLINEFORM3 are final encoding vectors from the audio-RNN and text-RNN, respectively  INLINEFORM4 and the bias INLINEFORM5 are learned model parameters 
Multimodal Dual Recurrent Encoder with Attention (MDREA)
Inspired by the concept of the attention mechanism used in neural machine translation BIBREF28 , we propose a novel multimodal attention method to focus on the specific parts of a transcript that contain strong emotional information, conditioning on the audio information  Figure shows the architecture of the MDREA model  First, the audio data and text data are encoded with the audio-RNN and text-RNN using equation EQREF2   We then consider the final audio encoding vector INLINEFORM0 as a context vector  As seen in equation EQREF9 , during each time step t, the dot product between the context vector e and the hidden state of the text-RNN at each t-th sequence INLINEFORM1 is evaluated to calculate a similarity score INLINEFORM2   Using this score INLINEFORM3 as a weight parameter, the weighted sum of the sequences of the hidden state of the text-RNN, INLINEFORM4 , is calculated to generate an attention-application vector Z  This attention-application vector is concatenated with the final encoding vector of the audio-RNN INLINEFORM5 (equation EQREF7 ), which will be passed through the softmax function to predict the emotion class  We use the same training objective as the ARE model, and the predicted probability distribution for the target class is as follows: DISPLAYFORM0
where INLINEFORM0 and the bias INLINEFORM1 are learned model parameters 
Dataset
We evaluate our model using the Interactive Emotional Dyadic Motion Capture (IEMOCAP) BIBREF18 dataset  This dataset was collected following theatrical theory in order to simulate natural dyadic interactions between actors

ground_truth: Unanswerable

Unanswerable

prompt_front: You are given a scientific article and a question. Answer the question as concisely as you can, using a single phrase or sentence if possible. If the question cannot be answered based on the information in the article, write "unanswerable". If the question is a yes/no question, answer "yes", "no", or "unanswerable". Do not provide any explanation.

Please answer the questions based on the text provided below. What were their results on the new dataset?
Context 2: 
In this work, we investigate how leveraging historical user preferences can improve generation quality over strong baselines in our setting  We compare our personalized models against two baselines  The first is a name-based Nearest-Neighbor model (NN)  We initially adapted the Neural Checklist Model of BIBREF0 as a baseline; however, we ultimately use a simple Encoder-Decoder baseline with ingredient attention (Enc-Dec), which provides comparable performance and lower complexity  All personalized models outperform baseline in BPE perplexity (tab:metricsontest) with Prior Name performing the best  While our models exhibit comparable performance to baseline in BLEU-1/4 and ROUGE-L, we generate more diverse (Distinct-1/2: percentage of distinct unigrams and bigrams) and acceptable recipes  BLEU and ROUGE are not the most appropriate metrics for generation quality  A `correct' recipe can be written in many ways with the same main entities (ingredients)  As BLEU-1/4 capture structural information via n-gram matching, they are not correlated with subjective recipe quality  This mirrors observations from BIBREF31, BIBREF8 
We observe that personalized models make more diverse recipes than baseline  They thus perform better in BLEU-1 with more key entities (ingredient mentions) present, but worse in BLEU-4, as these recipes are written in a personalized way and deviate from gold on the phrasal level  Similarly, the `Prior Name' model generates more unigram-diverse recipes than other personalized models and obtains a correspondingly lower BLEU-1 score 
Qualitative Analysis: We present sample outputs for a cocktail recipe in tab:samplerecipes, and additional recipes in the appendix  Generation quality progressively improves from generic baseline output to a blended cocktail produced by our best performing model  Models attending over prior recipes explicitly reference ingredients  The Prior Name model further suggests the addition of lemon and mint, which are reasonably associated with previously consumed recipes like coconut mousse and pork skewers 
Personalization: To measure personalization, we evaluate how closely the generated text corresponds to a particular user profile  We compute the likelihood of generated recipes using identical input specifications but conditioned on ten different user profiles—one `gold' user who consumed the original recipe, and nine randomly generated user profiles  Following BIBREF8, we expect the highest likelihood for the recipe conditioned on the gold user  We measure user matching accuracy (UMA)—the proportion where the gold user is ranked highest—and Mean Reciprocal Rank (MRR) BIBREF32 of the gold user  All personalized models beat baselines in both metrics, showing our models personalize generated recipes to the given user profiles  The Prior Name model achieves the best UMA and MRR by a large margin, revealing that prior recipe names are strong signals for personalization  Moreover, the addition of attention mechanisms to capture these signals improves language modeling performance over a strong non-personalized baseline 
Recipe Level Coherence: A plausible recipe should possess a coherent step order, and we evaluate this via a metric for recipe-level coherence  We use the neural scoring model from BIBREF33 to measure recipe-level coherence for each generated recipe  Each recipe step is encoded by BERT BIBREF34  Our scoring model is a GRU network that learns the overall recipe step ordering structure by minimizing the cosine similarity of recipe step hidden representations presented in the correct and reverse orders  Once pretrained, our scorer calculates the similarity of a generated recipe to the forward and backwards ordering of its corresponding gold label, giving a score equal to the difference between the former and latter  A higher score indicates better step ordering (with a maximum score of 2)  tab:coherencemetrics shows that our personalized models achieve average recipe-level coherence scores of 1 78-1 82, surpassing the baseline at 1 77 
Recipe Step Entailment: Local coherence is also crucial to a user following a recipe: it is crucial that subsequent steps are logically consistent with prior ones  We model local coherence as an entailment task: predicting the likelihood that a recipe step follows the preceding  We sample several consecutive (positive) and non-consecutive (negative) pairs of steps from each recipe  We train a BERT BIBREF34 model to predict the entailment score of a pair of steps separated by a [SEP] token, using the final representation of the [CLS] token  The step entailment score is computed as the average of scores for each set of consecutive steps in each recipe, averaged over every generated recipe for a model, as shown in tab:coherencemetrics 
Human Evaluation: We presented 310 pairs of recipes for pairwise comparison BIBREF8 (details in appendix) between baseline and each personalized model, with results shown in tab:metricsontest  On average, human evaluators preferred personalized model outputs to baseline 63% of the time, confirming that personalized attention improves the semantic plausibility of generated recipes  We also performed a small-scale human coherence survey over 90 recipes, in which 60% of users found recipes generated by personalized models to be more coherent and preferable to those generated by baseline models 
Conclusion
In this paper, we propose a novel task: to generate personalized recipes from incomplete input specifications and user histories  On a large novel dataset of 180K recipes and 700K reviews, we show that our personalized generative models can generate plausible, personalized, and coherent recipes preferred by human evaluators for consumption  We also introduce a set of automatic coherence measures for instructional texts as well as personalization metrics to support our claims  Our future work includes generating structured representations of recipes to handle ingredient properties, as well as accounting for references to collections of ingredients (e g  “dry mix") 
Acknowledgements  This work is partly supported by NSF #1750063  We thank all reviewers for their constructive suggestions, as well as Rei M , Sujoy P , Alicia L , Eric H , Tim S , Kathy C , Allen C , and Micah I  for their feedback 
Appendix ::: Food com: Dataset Details
Our raw data consists of 270K recipes and 1 4M user-recipe interactions (reviews) scraped from Food com, covering a period of 18 years (January 2000 to December 2018)  See tab:int-stats for dataset summary statistics, and tab:samplegk for sample information about one user-recipe interaction and the recipe involved 
Appendix ::: Generated Examples
See tab:samplechx for a sample recipe for chicken chili and tab:samplewaffle for a sample recipe for sweet waffles 
Human Evaluation
We prepared a set of 15 pairwise comparisons per evaluation session, and collected 930 pairwise evaluations (310 per personalized model) over 62 sessions  For each pair, users were given a partial recipe specification (name and 3-5 key ingredients), as well as two generated recipes labeled `A' and `B'  One recipe is generated from our baseline encoder-decoder model and one recipe is generated by one of our three personalized models (Prior Tech, Prior Name, Prior Recipe)  The order of recipe presentation (A/B) is randomly selected for each question  A screenshot of the user evaluation interface is given in fig:exeval  We ask the user to indicate which recipe they find more coherent, and which recipe best accomplishes the goal indicated by the recipe name  A screenshot of this survey interface is given in fig:exeval2 

ground_truth: average recipe-level coherence scores of 1.78-1.82, human evaluators preferred personalized model outputs to baseline 63% of the time

prompt_front: You are given a scientific article and a question. Answer the question as concisely as you can, using a single phrase or sentence if possible. If the question cannot be answered based on the information in the article, write "unanswerable". If the question is a yes/no question, answer "yes", "no", or "unanswerable". Do not provide any explanation.

Please answer the questions based on the text provided below. How do they combine audio and text sequences in their RNN?
Context 3:  We use categorical evaluations with majority agreement  We use only four emotional categories happy, sad, angry, and neutral to compare the performance of our model with other research using the same categories  The IEMOCAP dataset includes five sessions, and each session contains utterances from two speakers (one male and one female)  This data collection process resulted in 10 unique speakers  For consistent comparison with previous work, we merge the excitement dataset with the happiness dataset  The final dataset contains a total of 5531 utterances (1636 happy, 1084 sad, 1103 angry, 1708 neutral) 
Feature extraction
To extract speech information from audio signals, we use MFCC values, which are widely used in analyzing audio signals  The MFCC feature set contains a total of 39 features, which include 12 MFCC parameters (1-12) from the 26 Mel-frequency bands and log-energy parameters, 13 delta and 13 acceleration coefficients The frame size is set to 25 ms at a rate of 10 ms with the Hamming function  According to the length of each wave file, the sequential step of the MFCC features is varied  To extract additional information from the data, we also use prosodic features, which show effectiveness in affective computing  The prosodic features are composed of 35 features, which include the F0 frequency, the voicing probability, and the loudness contours  All of these MFCC and prosodic features are extracted from the data using the OpenSMILE toolkit BIBREF26  
Implementation details
Among the variants of the RNN function, we use GRUs as they yield comparable performance to that of the LSTM and include a smaller number of weight parameters BIBREF29   We use a max encoder step of 750 for the audio input, based on the implementation choices presented in BIBREF30 and 128 for the text input because it covers the maximum length of the transcripts  The vocabulary size of the dataset is 3,747, including the “_UNK_" token, which represents unknown words, and the “_PAD_" token, which is used to indicate padding information added while preparing mini-batch data  The number of hidden units and the number of layers in the RNN for each model (ARE, TRE, MDRE and MDREA) are selected based on extensive hyperparameter search experiments  The weights of the hidden units are initialized using orthogonal weights BIBREF31 ], and the text embedding layer is initialized from pretrained word-embedding vectors BIBREF32  
In preparing the textual dataset, we first use the released transcripts of the IEMOCAP dataset for simplicity  To investigate the practical performance, we then process all of the IEMOCAP audio data using an ASR system (the Google Cloud Speech API) and retrieve the transcripts  The performance of the Google ASR system is reflected by its word error rate (WER) of 5 53% 
Performance evaluation
As the dataset is not explicitly split beforehand into training, development, and testing sets, we perform 5-fold cross validation to determine the overall performance of the model  The data in each fold are split into training, development, and testing datasets (8:0 5:1 5, respectively)  After training the model, we measure the weighted average precision (WAP) over the 5-fold dataset  We train and evaluate the model 10 times per fold, and the model performance is assessed in terms of the mean score and standard deviation 
We examine the WAP values, which are shown in Table 1  First, our ARE model shows the baseline performance because we use minimal audio features, such as the MFCC and prosodic features with simple architectures  On the other hand, the TRE model shows higher performance gain compared to the ARE  From this result, we note that textual data are informative in emotion prediction tasks, and the recurrent encoder model is effective in understanding these types of sequential data  Second, the newly proposed model, MDRE, shows a substantial performance gain  It thus achieves the state-of-the-art performance with a WAP value of 0 718  This result shows that multimodal information is a key factor in affective computing  Lastly, the attention model, MDREA, also outperforms the best existing research results (WAP 0 690 to 0 688) BIBREF19   However, the MDREA model does not match the performance of the MDRE model, even though it utilizes a more complex architecture  We believe that this result arises because insufficient data are available to properly determine the complex model parameters in the MDREA model  Moreover, we presume that this model will show better performance when the audio signals are aligned with the textual sequence while applying the attention mechanism  We leave the implementation of this point as a future research direction 
To investigate the practical performance of the proposed models, we conduct further experiments with the ASR-processed transcript data (see “-ASR” models in Table )  The label accuracy of the processed transcripts is 5 53% WER  The TRE-ASR, MDRE-ASR and MDREA-ASR models reflect degraded performance compared to that of the TRE, MDRE and MDREA models  However, the performance of these models is still competitive; in particular, the MDRE-ASR model outperforms the previous best-performing model, 3CNN-LSTM10H (WAP 0 691 to 0 688) 
Error analysis
We analyze the predictions of the ARE, TRE, and MDRE models  Figure shows the confusion matrix of each model  The ARE model (Fig  ) incorrectly classifies most instances of happy as neutral (43 51%); thus, it shows reduced accuracy (35 15%) in predicting the the happy class  Overall, most of the emotion classes are frequently confused with the neutral class  This observation is in line with the findings of BIBREF30 , who noted that the neutral class is located in the center of the activation-valence space, complicating its discrimination from the other classes 
Interestingly, the TRE model (Fig  ) shows greater prediction gains in predicting the happy class when compared to the ARE model (35 15% to 75 73%)  This result seems plausible because the model can benefit from the differences among the distributions of words in happy and neutral expressions, which gives more emotional information to the model than that of the audio signal data  On the other hand, it is striking that the TRE model incorrectly predicts instances of the sad class as the happy class 16 20% of the time, even though these emotional states are opposites of one another 
The MDRE model (Fig  ) compensates for the weaknesses of the previous two models (ARE and TRE) and benefits from their strengths to a surprising degree  The values arranged along the diagonal axis show that all of the accuracies of the correctly predicted class have increased  Furthermore, the occurrence of the incorrect “sad-to-happy" cases in the TRE model is reduced from 16 20% to 9 15% 
Conclusions
In this paper, we propose a novel multimodal dual recurrent encoder model that simultaneously utilizes text data, as well as audio signals, to permit the better understanding of speech data  Our model encodes the information from audio and text sequences using dual RNNs and then combines the information from these sources using a feed-forward neural model to predict the emotion class  Extensive experiments show that our proposed model outperforms other state-of-the-art methods in classifying the four emotion categories, and accuracies ranging from 68 8% to 71 8% are obtained when the model is applied to the IEMOCAP dataset  In particular, it resolves the issue in which predictions frequently incorrectly yield the neutral class, as occurs in previous models that focus on audio features 
In the future work, we aim to extend the modalities to audio, text and video inputs  Furthermore, we plan to investigate the application of the attention mechanism to data derived from multiple modalities  This approach seems likely to uncover enhanced learning schemes that will increase performance in both speech emotion recognition and other multimodal classification tasks 
Acknowledgments
K  Jung is with the Department of Electrical and Computer Engineering, ASRI, Seoul National University, Seoul, Korea  This work was supported by the Ministry of Trade, Industry & Energy (MOTIE, Korea) under Industrial Technology Innovation Program (No 10073144) 

ground_truth: combines the information from these sources using a feed-forward neural model

encodes the information from audio and text sequences using dual RNNs and then combines the information from these sources using a feed-forward neural model