I NTO THE W ILD WITH A UDIO S COPE : U NSUPERVISED A UDIO -V ISUAL S EPARATION OF O N -S CREEN S OUNDS
Anonymous authors
Paper under double-blind review
A BSTRACT
Recent progress in deep learning has enabled many advances in sound separation and visual scene understanding.
However, extracting sound sources which are apparent in natural videos remains an open problem.
In this work, we present
AudioScope , a novel audio-visual sound separation framework that can be trained without supervision to isolate on-screen sound sources from real in-the-wild videos.
Prior audio-visual separation work assumed artiﬁcial limitations on the domain of sound classes (e.g., to speech or music), constrained the number of sources, and required strong sound separation or visual segmentation labels.
AudioScope overcomes these limitations, operating on an open domain of sounds, with variable numbers of sources, and without labels or prior visual segmentation.
The training procedure for AudioScope uses mixture invariant training (MixIT) to separate synthetic mixtures of mixtures (MoMs) into individual sources, where noisy labels for mixtures are provided by an unsupervised audio-visual coincidence model.
Using the noisy labels, along with attention between video and audio features,
AudioScope learns to identify audio-visual similarity and to suppress off-screen sounds.
We demonstrate the effectiveness of our approach using a dataset of video clips extracted from open-domain YFCC100m video data.
This dataset contains a wide diversity of sound classes recorded in unconstrained conditions, making the application of previous methods unsuitable.
For evaluation and semi-supervised experiments, we collected human labels for presence of on-screen and off-screen sounds on a small subset of clips.
I NTRODUCTION
[Figure]
Audio-visual machine perception has been undergoing a renaissance in recent years driven by advances in large-scale deep learning.
A motivating observation is the interplay in human perception between auditory and visual perception.
We understand the world by parsing it into the objects that are the sources of the audio and visual signals we can perceive.
However, the sounds and sights produced by these sources have rather different and complementary properties.
Objects may make sounds intermittently, whereas their visual appearance is typically persistent.
The visual percepts of different objects tend to be spatially distinct, whereas sounds from different sources can blend together and overlap in a single signal, making it difﬁcult to separately perceive the individual sources.
This suggests that there is something to be gained by aligning our audio and visual percepts: if we can identify which audio signals correspond to which visual objects, we can selectively attend to an object’s audio signal by visually selecting the object.
This intuition motivates using vision as an interface for audio processing, where a primary problem is to selectively preserve desired sounds, while removing unwanted sounds.
In some tasks, such as speech enhancement, the desired sounds can be selected by their class: speech versus non-speech in this case.
In an open-domain setting, the selection of desired sounds is at the user’s discretion.
This presents a user-interface problem: it is challenging to select sources in an efﬁcient way using audio.
This problem can be greatly simpliﬁed in the audio-visual case if we use video selection as a proxy for audio selection, for example, by selecting sounds from on-screen objects, and removing off-screen sounds.
Recent work has used video for selection and separation of speech (Ephrat et al., 2018;
Afouras et al., 2020) or music (Zhao et al., 2018; Gao & Grauman, 2019; Gan et al., 2020).
However, systems that address this for arbitrary sounds (Gao et al., 2018; Rouditchenko et al., 2019;
Owens & Efros, 2018) may be useful in more general cases, such as video recording, where the sounds of interest cannot be deﬁned in advance.
The problem of associating arbitrary sounds with their visual objects is challenging in an open domain.
Several complications arise that have not been fully addressed by previous work.
First, a large amount of training data is needed in order to cover the space of possible sound.
Supervised methods require labeled examples where isolated on-screen sounds are known.
The resulting data collection and labeling burden limits the amount and quality of available data.
To overcome this, we propose an unsupervised approach using mixture invariant training (MixIT)
(Wisdom et al., 2020b), that can learn to separate individual sources from in-the-wild videos, where the on-screen and off-screen sounds are unknown.
Another problem is that different audio sources may correspond to a dynamic set of on-screen objects in arbitrary spatial locations.
We accommodate this by using attention mechanisms that align each hypothesized audio source with the different spatial and temporal positions of the corresponding objects in the video.
Finally we need to determine which audio sources appear on screen, in the absence of strong labels.
This is handled using a weakly trained classiﬁer for sources based on audio and video embeddings produced by the attention mechanism.
R ELATION TO PREVIOUS WORK
Separation of arbitrary sounds from a mixture, known as “universal sound separation,” was recently shown to be possible with a ﬁxed number of sounds (Kavalerov et al., 2019).
Conditional information about which sound classes are present can improve separation performance (Tzinis et al., 2020).
The
FUSS dataset (Wisdom et al., 2020a) expanded the scope to separate a variable number of sounds, in order to handle more realistic data.
A framework has also been proposed where speciﬁc sound classes can be extracted from input sound mixtures (Ochiai et al., 2020).
These approaches require curated data containing isolated sounds for training, which prevents their application to truly open-domain data and introduces difﬁculties such as annotation cost, accurate simulation of realistic acoustic mixtures, and biased datasets.
To avoid these issues, a number of recent works have proposed replacing the strong supervision of reference source signals with weak supervision labels from related modalities such as sound class (Pishdadian et al., 2019; Kong et al., 2020), visual input (Gao & Grauman, 2019), or spatial location from multi-microphone recordings (Tzinis et al., 2019; Seetharaman et al., 2019; Drude et al., 2019).
Most recently, Wisdom et al.
(2020b) proposed mixture invariant training (MixIT), which provides a purely unsupervised source separation framework for a variable number of latent sources.
A variety of research has laid the groundwork towards solving audio-visual on-screen source separation (Michelsanti et al., 2020).
Generally, the two main approaches are to use audio-visual localization (Hershey & Movellan, 2000; Senocak et al., 2018;
Wu et al., 2019; Afouras et al., 2020), or object detection networks, either supervised (Ephrat et al., 2018; Gao & Grauman, 2019; Gan et al., 2020) or unsupervised (Zhao et al., 2018), to predict visual conditioning information.
However, these works only consider restricted domains such as speech (Hershey & Casey, 2002;
Ephrat et al., 2018; Afouras et al., 2020) or music (Zhao et al., 2018; Gao & Grauman, 2019; Gan et al., 2020).
Gao et al.
(2018) reported results with videos from a wide domain, but relied on supervised visual object detectors, which precludes learning about the appearance of sound sources outside of a closed set of classes deﬁned by the detectors.
Rouditchenko et al.
(2019) proposed a system for a wide domain of sounds, but required sound class labels as well as isolated sounds from these classes.
Our approach avoids the supervision of class labels and isolated sources in order to handle unknown visual and sound classes occurring in multi-source data.
Towards learning directly from a less restrictive open domain of in-the-wild video data, Tian et al.
(2018) learn to localize audio-visual events in unconstrained videos and present an ad hoc dataset (Korbar et al., 2018) that pre-trains models to discern temporal synchronization of audio-video pairs, and demonstrate promising results on action recognition and audio classiﬁcation.
Arandjelovic &
Zisserman (2017) take a similar approach by classifying audio-visual correspondences of pairs of one video frame and one second of audio.
Hu et al.
(2020) proposed a curriculum learning approach where the model gradually learns harder examples to separate.
Closest to our work is the approach of Owens & Efros (2018), a self-supervised audio-visual onscreen speech separation system based on temporal audio-visual alignment.
However, Owens &
Efros (2018) assumes training videos containing only on-screen sources, and it is unclear how to adapt it to the case where training videos include off-screen sources.
Our approach signiﬁcantly differs from these prior works in that we do not restrict our domain to musical instruments or human speakers, and we train and test with real in-the-wild videos containing an arbitrary number of objects with no object class restrictions.
Our proposed framework can deal with noisy labels (e.g. videos with no on-screen sounds), operate on a completely open-domain of in-the-wild videos, and effectively isolate sounds coming from on-screen objects.
We address the following task, which extends the formulation of the on-screen speech separation problem (Owens & Efros, 2018).
Given an input video, the goal is to separate all sources that constitute the input mixture, and then estimate an audio-visual correspondence score for each separated source.
These probability scores should be high for separated sources which are apparent on-screen, and low otherwise.
The separated audio sources, weighted by their estimated on-screen probabilities, can be summed together to reconstruct the on-screen mixture.
We emphasize that our approach is more generally applicable than previous proposals, because real-world videos may contain an unknown number of both on-screen and off-screen sources belonging to an undeﬁned ontology of classes.
We make the following contributions in this paper:
We provide the ﬁrst solution for training an unsupervised, open-domain, audio-visual onscreen separation system from scratch on real in-the-wild video data, with no requirement on modules such as object detectors that require supervised data.
We provide a new dataset for the on-screen audio-visual separation task, consisting of 2 ,hours of unlabeled videos from YFCC100m, and 55 hours of videos that are human-labeled for presence of on-screen and off-screen sounds.
M ODEL ARCHITECTURE
The overall architecture of AudioScope is built from the following blocks: an image embedding network, an audio separation network, an audio embedding network, an audio-visual attention mechanism, and an on-screen classiﬁer (see Figure 2). The separation and embedding networks are based on prior work and are described in the following subsections. However, the main focus of this work is the overall architecture, as well as the training framework and loss functions. The video is analyzed with the image embedding network, which generates local embeddings for each of 64 locations within each frame, as well as an embedding of the whole frame.
These embeddings are used both as a conditioning input to an audio separation network, as well as an input for classiﬁcation of the on-screen sounds. The audio separation network takes the mixed input waveform as input, and generates a ﬁxed number of output waveforms, a variable number of which are non-zero depending on the estimated number of sources in the mixture. Conditioning on the video enables the separation to take advantage of cues about the sources present when performing separation. The audio embedding network is applied to each estimated source to obtain one embedding per frame for each source.
These audio embeddings are then pooled over time and used in the audio-visual spatio-temporal attention network to retrieve, for each source, a representation of the visual activity that best matches the audio, similar to the associative maps extracted from the internal network representations proposed by Harwath et al.
The architecture is designed to address the problem of unsupervised learning on in-the-wild opendomain data.
First, because the target training videos can contain both on-screen and off-screen
[Figure]
Spatiotemporalattention
[Figure]
Temporalpooling
On-screen classiﬁer
[Figure]
[Figure]
embeddingnetwork
Predictions
Pseudo
[Figure]
sounds, training a system to directly produce the audio of the target video would encourage inclusion of off-screen sounds as well as on-screen ones 1 .
Our proposed multi-source separation network instead produces latent source estimates using an unsupervised MixIT objective, which has been shown to perform well at general sound separation Wisdom et al.
(2020b).
By decoupling separation from on-screen classiﬁcation, our architecture facilitates the use of robust objectives that allow some of the sources to be considered off-screen, even if they appear in the soundtrack of the target videos.
The audio-visual attention architecture is motivated by the alignment problem between audio and video: sound source objects in video may be localized, may move over time, and may be present before and after the corresponding audio activity.
Because of the open domain we cannot rely on a pre-deﬁned set of object detectors to anchor the video representations of on-screen sources, as is done in some prior works (Ephrat et al., 2018; Gao & Grauman, 2019; Gan et al., 2020).
Instead we propose attention to ﬁnd the video representations that correspond to a source in a more ﬂexible way.
The proposed strategy of temporal pooling of the audio embeddings, before using them in the spatiotemporal attention, allows the network to derive embeddings that represent the active segments of the source audio, and ignore the ambiguous silent regions.
In the present model, video is analyzed at a low frame rate, and so the audio-visual correspondence is likely based on relatively static properties of the objects, rather than the synchrony of their motion with the audio.
In this case, a single time-invariant representation of the audio may be sufﬁcient as a proof of concept.
However, in future work, with higher video frame rates, it may be worthwhile to consider using attention to align sequences of audio and video embeddings in order to detect synchrony in their activity patterns.
The on-screen classiﬁer operates on an audio embedding for one estimated source, as well as the video embedding retrieved by the spatio-temporal attention mechanism, using a dense network.
This presumably allows detection of the congruence between the embeddings.
To provide additional context for this decision, a global video embedding, produced by temporal pooling, is provided as an additional input. Many alternative choices are possible for this classiﬁer design, which we leave for future work, such as using a more complex classiﬁcation architecture, or providing additional audio embeddings as input. 3. A UDIO SEPARATION NETWORK The separation network M s architecture consists of learnable convolutional encoder and decoder layers with an improved time-domain convolutional network (TDCN++) masking network (Wisdom et al., 2020b).
A mixture consistency projection (Wisdom et al., 2019) is applied to constrain separated sources to add up to the input mixture.
The separation network processes a T -sample input mixture waveform and outputs M estimated sources ˆ s ∈ R M × T .
Internally, the network estimates M masks which are multiplied with the activations of the encoded input mixture.
The time-domain signals ˆ s are computed by applying the decoder, a transposed convolutional layer, to the masked coefﬁcients.
3. A UDIO EMBEDDING NETWORK
For each separated source ˆ s m , we extract a corresponding global audio embedding using the MobileNet v1 architecture (Howard et al., 2017) which consists of stacked 2D separable dilated convolutional blocks with a dense layer at the end.
This network M (cid:97) ﬁrst computes log Mel-scale spectrograms with F a audio frames from the time-domain separated sources, and then applies stacks of depthwise separable convolutions to produce the F a × N embedding matrix Z a m , which contains an N -dimensional row embedding for each frame.
An attentional pooling operation (Girdhar & Ramanan, 2017) is used, for each source, m , to form a static audio embedding vector z a m = attend( ¯ Z a m ,
[-On-screen Classifier 1 2 31 2 
 [Figure] row embedding for each frame. An attentional pooling operation (Girdhar & Ramanan, 2017) is used, for each source m , to form an embedding vector-] Z a m , Z a m ) , where the average embedding ¯ Z a m = 1 F a (cid:80) i Z a m,i is the query vector for source m .
The attention mechanism (Bahdanau et al., 2014) is deﬁned as follows:
[Equation]
with query row vector q , the attention weight distribution column vector α , key matrix K , value matrix V , and trainable row-wise dense layers f q , f V , f K , all having conforming dimensions.
3. I MAGE EMBEDDING NETWORK
To extract visual features from video frames, we again use a MobileNet v1 architecture.
This visual embedding model M v is applied independently to each one of the F v input video frames and a static-length embedding is extracted for each image Z v j , j ∈ { 1 , . .
, F v } .
Conditioning separation network with the temporal video embedding : The embeddings of the video input Z v j can be used to condition the separation network (Tzinis et al., 2020). Speciﬁcally, the image embeddings are fed through a dense layer, and a simple nearest neighbor upsampling matches the time dimension to the time dimension of the intermediate separation network activations.
These upsampled and transformed image embeddings are concatenated with the intermediate TDCN++ activations and fed as input to the separation network layers.
Global video embedding : A global embedding of the video input is extracted using attentional pooling over all video frames, given by z vg = attend( ¯ Z v , Z v , Z v ) , where the average embedding ¯
Z v = 1 F v (cid:80) j Z v j is the query vector.
Local spatio-temporal video embedding : We also use local features extracted from an intermediate level in the visual convolutional network, that has 8 × 8 spatial locations.
These are denoted Z vl k , where k =
( j, n ) indexes video frame j and spatial location index n .
These provide spatial features for identiﬁcation of sources with visual objects to be used with audio-visual spatio-temporal attention.
3. A UDIO - VISUAL SPATIO - TEMPORAL ATTENTION
An important aspect of this work is to combine audio and visual information in order to infer correspondence between each separated source and the relevant objects in video.
This in turn will be used to identify which sources are visible on-screen.
To this end, we employ an audio-visual spatiotemporal attention scheme by letting the network attend to the local features of the visual embeddings for each separated source.
In this mechanism, we use the audio embedding Z a m as the query input for source m , and the key and value inputs are given by the spatio-temporal video embeddings, Z vl .
As a result, the ﬂattened version of the output spatio-temporal embedding, corresponding to the m -th source, is z av m = attend( z a m , Z vl , Z vl ) .
3. O N - SCREEN C LASSIFIER
To infer the visual presence each separated source, we concatenate the global video embedding z vg , the global audio embedding for each source z a m , and the corresponding local spatio-temporal audio-visual embedding z av m .
The concatenated vector is fed through a dense layer with a logistic activation:
ˆ y m = logistic ([ z vg , z a m , z av m ]) .
T RAINING L OSSES 4. S EPARATION L OSS
We use a MixIT separation loss (Wisdom et al., 2020b), which optimizes the assignment of M estimated sources ˆ s =
M s ( x 1 + x 2 ) to two reference mixtures x 1 , x 2 as follows:
[Equation]
where the mixing matrix A ∈ B 2 × M is constrained to the set of 2 × M binary matrices where each column sums to 1 and the SNR loss is deﬁned as
[Equation]
Due to the constraints on A , each source ˆ s m can only be assigned to one reference mixture.
4. C LASSIFICATION L OSS
To train the on-screen classiﬁer, we consider the following classiﬁcation losses.
These losses use the binary labels y m , where are given for supervised examples, and in the unsupervised case y m =
A ∗ 0 ,m for each source m , where A ∗ is the optimial mixing matrix found by the minimization in (2). We also use the notation R = { m | y m = 1 , m ∈ { 1 , . . . , M }} to denote the set of positive labels. Exact binary cross entropy :
[Equation]
Multiple-instance cross entropy : Since some separated sources assigned to the on-screen mixture are not on-screen, a multiple-instance (MI) (Maron & Lozano-Pérez, 1998) loss, which minimizes over the set of positive labels R may be more robust:
[Equation]
Active combinations cross entropy : An alternative to the MI loss, active combinations (AC), corresponds to the minimum loss over all settings ℘ ≥ 1 ( R ) of the labels s.t. at least one label is 1 :
[Equation]
where ℘ ≥ 1 ( R ) denotes the power set of indices with label of y i = 1 .
E XPERIMENTAL F RAMEWORK 5. D ATA PREPARATION
In order to train on real-world audio-visual recording environments for our open-domain system, we use the Yahoo Flickr Creative Commons 100 Million Dataset (YFCC100m)
(Thomee et al., 2016).
The dataset consists of about 200 , 000 videos ( 2 , 500 hours) of various lengths and covering a diverse range of semantic sound categories.
We extract 5-second clips with a hop size of 1 second, resulting in around 7 . 2 million clips.
Clips consist of a 5-second audio waveform sampled at 16 kHz andvideo frames x ( f ) , where each frame is a 128 × 128 × 3 RGB image.
Our goal is to train our system completely unsupervised, but we sought to reduce the proportion of videos where there are no on-screen sounds.
We thus created a ﬁltered subset D f of YFCC100m of clips with a high audio-visual coincidence probability predicted by an unsupervised audio-visual coincidence prediction model (Jansen et al., 2020) trained on sounds from AudioSet (Gemmeke et al., 2017).
The resulting selection is noisy, because the coincidence model is not perfect and clips that have high audio-visual coincidence may contain both on-screen and off-screen sounds, or even no on-screen sounds, but at least improves the presence of on-screen sounds, as shown below.
The ﬁnal ﬁltered dataset consists of all clips (about 336 , 000 ) extracted from the 36 , 000 highest audio-visual coincidence scoring videos.
The threshold for ﬁltering was empirically set to keep a fair amount of diverse videos while ensuring that not too many clips with only off-screen sounds were accepted.
In order to evaluate the performance of the unsupervised ﬁltering as well as our proposed models, and to experiment with a small amount of supervised data, we randomly selected 20 , 000 unﬁltered video clips and 20 , 000 ﬁltered clips to obtain human annotations.
In the annotation process, the raters chose whether on-screen or off-screen sounds are present or not present.
Each clip is labeled by 3 individual raters, and a clip is only considered on-screen or off-screen only if all 3 raters are unanimous.
We constructed an on-screen subset with 836 clips for training, 735 for validation, and 295 for test, and an off-screen subset with 3 , 681 clips for training, 836 for validation, and 370 for test.
Based on human annotations, we estimate that for unﬁltered data 71 .
3% of clips contain both onand-off-screen sounds, 2 . 8% contain on-screen-only sounds, and 25 . 9% contain off-screen-only.
For the ﬁltered data, 83 .
5% of clips contain on-screen and off-screen sounds, 5 .
6% of clips are on-screen-only, and 10 .
9% are off-screen-only.
Thus the unsupervised ﬁltering signiﬁcantly reduced the proportion of off-screen-only clips.
5. T RAINING
Both audio and visual embedding networks were pre-trained on Audioset (Gemmeke et al., 2017) for unsupervised coincidence prediction (Jansen et al., 2020) and ﬁne-tuned on our data (see Appendix A.3.1 for ablation), whereas the separation network is trained from scratch using MixIT (2) on mixtures of mixtures (MoMs) constructed from the audio of our data.
All models are trained on Google Cloud TPUs (16 chips) with the Adam optimizer (Kingma & Ba, 2015), a batch size of 256 , and learning rate of 10 − 4 .
In order to train the overall network, we use 2 , 500 hours of unlabeled video-clips and construct minibatches of video clips with MoMs as audio.
The ﬁrst mixture in these MoMs is one of four types:
1. NOn: noisy-labeled on-screen.
SOff: synthetic off-screen-only.
LOn: human-labeled on-screen-only.
LOff: human-labeled off-screen-only.
For all example types, the second mixture in the MoM is audio drawn from a different random video in the unﬁltered data.
A NOn MoM uses video clip frames and audio from the ﬁltered high-coincidence subset of our data, D f .
SOff MoMs are constructed by combining video frames of an unﬁltered clip with random audio drawn from a different unﬁltered video.
Unsupervised training minibatches consist of either 0% or 25% SOff examples, and the remaining examples are NOn.
Semi-supervised minibatches additionally include LOn and LOff examples, which are constructed in the manner as NOn, except that the ﬁrst mixture is drawn from unanimously human-labeled on-screenonly videos and unanimously human-labeled off-scren-only videos, respectively.
We experiment with using 0% or 25% SOff examples, which corresponds to minibatch (NOn, SOff, LOn, LOff) proportions of ( 50% , 0% , 25% , 25% ) and ( 25% , 25% , 25% , 25% ), respectively.
Classiﬁcation labels y m for all separated sources ˆ s m in SOff and LOff examples are set to 0.
For
NOn and LOn examples, we set the label for each separated source as y m =
A 0 ,m , where A is the
MixIT assignment matrix (2).
The MixIT separation loss (2) is used for all MoM example types.
5. E VALUATION
All evaluations use human-labeled test videos, which have been unanimously labeled as containing either only on-screen or only off-screen sounds.
Using this data, we construct four evaluation sets: on-screen single mixtures, off-screen single mixtures, on-screen MoMs, and off-screen MoMs.
The single mixture evaluations consist of only data drawn from the particular label, either on-screen or off-screen.
Each on-screen (off-screen) MoM consists of an on-screen-only (off-screen-only) video clip, mixed with the audio from another random clip, drawn from the off-screen-only examples.
5.3. O N - SCREEN D ETECTION
Detection performance for the on-screen classiﬁer is measured using the area under the curve of the receiver operator characteristic (AUC-ROC).
To avoid ambiguous classiﬁcation decisions due to inactive or very quiet separated sources, we use a weight for each source signal’s prediction equal to the linear power ratio between the powers of the source and input MoM. For single mixture evaluations, positive labels are assumed for all separated sources that are estimated from on-screen-only mixtures, and negative labels for all separated sources from off-screen-only mixtures.
For on-screen MoM evaluations, labels for separated sources from on-screen MoMs are assigned using the ﬁrst row of the oracle MixIT mixing matrix, classiﬁer outputs for sources separated from off-screen MoMs are assigned negative labels.
5.3. S EPARATION
Since we do not have access to individual ground-truth reference sources for our in-the-wild data, we cannot evaluate the per-source separation performance.
The only references we have are mixtures.
Thus, we compute an estimate of the on-screen audio by combining the separated sources using classiﬁer predictions: ˆ x on = (cid:80) Mm =1 p m ˆ s m .
For on-screen single mixture and MoM evaluations, we measure scale-invariant SNR (SI-SNR) (Le Roux et al., 2019), between ˆ x on and the reference on-screen-only mixture x
( on ) .
SI-SNR measures the ﬁdelity between a target y and an estimate ˆ y within an arbitrary scale factor in units of decibels:
[Equation]
To measure the degree to which AudioScope rejects off-screen audio, we deﬁne the off-screen suppression ratio (OSR), which is the ratio in decibels of the power of the input mixture to the power of the on-screen estimate ˆ x on .
We only compute OSR for off-screen evaluation examples where the input mixture only contains off-screen audio.
Thus, higher OSR implies greater suppression of off-screen sounds.
The minimum value of OSR is 0 dB, which means that ˆ x on is equal to the input mixture, which corresponds to all on-screen classiﬁer probabilities equal to 1.
In some cases, SI-SNR and OSR can yield inﬁnite values.
For example, the estimate ˆ y may be zero, in which case SI-SNR (7) is −∞ dB. This can occur when the input SNR of an on-screen mixture in a MoM is very low and none of the separated sources are assigned to it by MixIT.
Conversely, if the estimate perfectly matches the target, SI-SNR can yield a value of ∞ dB, which occurs for on-screen single mixture evaluation cases when the separated sources trivially add up to the on-screen input due to mixture consistency of the separation model.
For off-screen examples, OSR can also be inﬁnite if the separation model achieves perfect off-screen suppression by predicting zero for ˆ x on .
To avoid including these inﬁnite values, we elect to measure median SI-SNR and OSR.
R ESULTS
Results are shown in Table 1.
Note that there is a trade-off between preservation of on-screen sounds, as measured by SI-SNR, and suppression of off-screen sounds, as measured by OSR: higher on-screen SI-SNR on on-screen examples generally means lower OSR on off-screen examples.
Different classiﬁcation losses have different operating points: for MoMs, compared to using the exact cross-entropy loss, models trained with active combinations or multiple instance loss achieve lower on-screen SI-SNR, while achieving more suppression (higher OSR) of off-screen sounds.
Exact cross-entropy models achieve higher AUC-ROC for single mixtures and MoMs, and achieve better reconstruction of on-screen single mixtures at the expense of less rejection of off-screen mixtures.
Training only with the noisy labels provided by the unsupervised coincidence model (Jansen et al., 2020) achieves lower AUC-ROC compared to the semi-supervised condition that adds a small amount of human-labeled examples.
Semi-supervised and unsupervised models achieve comparable onscreen SI-SNR, but semi-supervised models achieve better off-screen suppression.
For example, the best on-screen SI-SNR for unsupervised and semi-supervised is 8.0 dB and 7.3 dB, respectively, while OSR is 5.3 dB and 10.7 dB. Using 25% synthetic off-screen particularly shifts the behavior of semi-supervised models by biasing them towards predicting lower probabilities of on-screen. This bias results in lower on-screen SI-SNR, yet very strong off-screen rejection (i.e. very large OSRs).
[Table]
MI 0. ∞ 12. 21.0 Semi 0%
Exact 0.
∞ 32. 10.7 Semi 25% AC 0. ∞ 6.
61.8 Semi 25%
MI 0.
∞ 6.
60.1 Semi 25%
Exact 0. ∞ 6. Figure 3 shows scatter plots of input SI-SNR versus SI-SNR of MixIT or ˆ x on on-screen estimates. From these plots, it is clear that the models tend to improve on-screen SI-SNR more often than not, and that these improvements are most signiﬁcant around ± 10 dB input SI-SNR.
Note that for MixIT, a number of points have a SI-SNR of −∞ , which happens when MixIT assigns all separated sources to the off-screen mixture.
OSR is sometimes ∞ when AudioScope achieves excellent off-screen suppression, predicting nearly 0 for the on-screen audio from off-screen-only input.
To provide a sense of the qualitative performance of AudioScope, we include visualizations of best, worst, and typical predictions in the appendix, and the supplementary material contains audio-visual demos. To benchmark AudioScope against other audio-visual separation approaches and measure performance on mismatched data, we evaluate on existing audio-visual separation test sets in Appendix A.2. We also performed a number of ablations for AudioScope, described in Appendix A.3.
C ONCLUSION
In this paper we have proposed the ﬁrst solution for training an unsupervised, open-domain, audiovisual on-screen separation system, without reliance on prior class labels or classiﬁers.
We demonstrated the effectiveness of our system using a small amount of human-labeled, in-the-wild videos, which we also plan to release to the community.
In future work, we will explore more ﬁne-grained visual features, especially synchrony, which we expect will be especially helpful when multiple instances of the same object are present in the video.
We also plan to use our trained classiﬁer to reﬁlter YFCC100m to get better noisy labels for the presence of on-screen sounds, which should further improve the performance of the system.
A A PPENDIX A. Q UALITATIVE EXAMPLES
For a range of input SNRs, Figure 4 shows best-case examples of separating on-screen sounds with
AudioScope, while Figure 5 shows failure cases.
Figures 6 and 7 show random examples at various
SNRs, comparing the outputs of semi-supervised SOff 0% models trained with either exact cross entropy (4) or active combinations cross entropy (6).
Figure 6 shows the outputs of the two models on 7 random examples, and Figure 7 shows the outputs of the two models on 5 examples that have maximum absolute difference in terms of SI-SNR of the on-screen estimate.
The supplementary material includes audio-visual demos of AudioScope on single mixtures and
MoMs with visualizations of MixIT assignments and predicted on-screen probabilities.
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A. E VALUATION ON MISMATCHED DATA
To evaluate the generalization capability of AudioScope and facilitate a comparison to prior works, we evaluated our model using test data from an audio-visual speech enhancement task (Hou et al., 2018) as well as an audio-visual task derived from a single-source subset of AudioSet (Gao et al., 2018).
In both cases, the evaluation is on a restricted domain, and the prior methods used both matched and supervised training on that domain.
In contrast, the AudioScope model is trained on open-domain YFCC100m videos using unsupervised training.
For all evaluations we use the unsupervised AudioScope model using 0% SOff and active combinations loss.
A.2. E VALUATION ON A UDIO -V ISUAL S PEECH E NHANCEMENT
Since our method can be used to separate on-screen sounds for arbitrary classes of sound, to compare to existing approaches we evaluate the trained AudioScope model on the more restricted domain of audio-visual speech enhancement.
To that end, we used the Mandarin sentences dataset, introduced by Hou et al.
The dataset contains video utterances of Mandarin sentences spoken by a native speaker.
Each sentence is unique and contains 10 Chinese characters.
The length of each utterance is approximately 3 to 4 seconds.
Synthetic noise is added to each ground truth audio.
Forty such videos are used as the ofﬁcial testing set.
For our evaluation we regard the speech of the ﬁlmed speaker as the on-screen sounds and the interference as off-screen sounds.
Thus, we can compute quality metrics for the on-screen estimate while comparing to speech enhancement methods.
To compare to previously-published numbers, we use signal-to-distortion ratio (SDR) (Vincent et al., 2006), which measures signal-to-noise ratio within a linear ﬁltering of the reference signal.
Table 2 shows the comparison between Hou et al.
(2018), Ephrat et al.
AudioScope ˆ x on (on-screen estimate using predicted on-screen probabilities), AudioScope source with max ˆ y m (use separated source with highest predicted on-screen probability),
AudioScope best source (oracle selection of the separated source with the highest SDR with on-screen reference), and AudioScope
MixIT* (on-screen estimate using oracle binary on-screen weights using references).
Note that the
AudioScope models are trained with mismatched open-domain training data, whereas the others were trained on matched speech enhancement data.
It can be seen that although non-oracle AudioScope estimates do not advance on state-of-the-art performance of speech enhancement-speciﬁc methods, the oracle AudioScope estimates improve over (Hou et al., 2018).
Thus AudioScope show promising results on this challenging data which is not explicitly represented in its open-domain training set.
We believe that by adding such data to our training set, perhaps by ﬁne-tuning, AudioScope could improve its performance signiﬁcantly on this more speciﬁc task, which we leave for future work
[Table]
[Table]
A.2. E VALUATION ON A UDIO S ET -S INGLE S OURCE
We evaluated AudioScope on the musical instrument portion of the AudioSet-SingleSource dataset (Gao et al., 2018), which is a small number of clips from AudioSet (Gemmeke et al., 2017) that have been veriﬁed by humans to contain single sources.
We use the same procedure as Gao & Grauman (2019) to construct a MoM test set, which creates 105 synthetic mixtures from all pairs of 15 musical instrument classes.
For each pair, audio tracks are mixed together, and we perform separation twice for each pair, conditioning on the video for each source.
The results are shown in table 3.
The non-oracle AudioScope methods perform rather poorly, but the oracle methods, especially
MixIT* (which matches the MixIT training loss), achieve state-of-the-art performance compared to
[Table]
methods form the literature.
This suggests that the on-screen classiﬁer is less accurate on this data.
Also, mixing the predicted AudioScope sources using the probabilities of the on-screen classiﬁer may be suboptimal
[Table]
and exploring alternative mixing methods to estimate on-screen audio is an avenue for future work.
Fine-tuning on data for this speciﬁc task could also improve performance, which we also leave for future work.
A. A BLATIONS
We performed a number of ablations on AudioScope.
The following subsections show the results of a number of ablations using either unsupervised or semi-supervised training. All models for these ablation use 0% SOff examples and the active combinations loss (6). A.3. A UDIO AND VIDEO EMBEDDINGS Table 4 shows the results of various ablations involving audio and video embeddings in the model.
related to audio and video embeddings
[Table]
First
[Table]
notice that removing video conditioning for the separation model reduces on-screen SI-SNR by 2 dB on single mixtures and 0.9 dB on MoMs, with negligible or slight improvement in OSR.
Thus, we can conclude that visual conditioning does have some beneﬁt for the model.
Next, we consider training the audio and video embedding networks from scratch, instead of using the coincidence model weights pretrained using AudioSet (Jansen et al., 2020).
Training from scratch is quite determinental, as AUC-ROC decreases by a minimum of 0.13 and maximum of 0.
across single-mixtures/MoMs and unsupervised/semi-supervised conditions.
Furthermore, separation performance suffers, with on-screen SI-SNR dropping by multiple for all conditions.
Finally, we consider removing the global video embedding, or both the global video embedding and audio embeddings, from the input of the on-screen classiﬁer.
This results in equivalent or slightly worse AUC-ROC, with equivalent or worse on-screen SI-SNR.
For unsupervised training, removing both embeddings at the classiﬁer input improves on-screen SI-SNR a bit (0.5 dB for single mixtures, 0.6 dB for MoMs) with a slight drop in OSR, though for semi-supervised on-screen SI-SNR drops by
3.7 dB for single mixtures and 0.5 dB for MoMs.
Overall, the best result is achieved by including these embeddings at the classiﬁer input
[Table]
A.3. A TTENTIONAL POOLING
We tried decreasing the embedding dimension from 256 to 128, as well as replacing the attentional pooling with mean pooling for audio sources, video frames, or both.
The results are shown in Table 5.
[Table]
Decreasing the embedding dimension reduces performance, dropping on-screen SI-SNR by 1.4 dB on single mixtures and 0.6 dB on MoMs, also with reduction in OSR.
Replacing attentional pooling with mean pooling generally does not change AUC-ROC or on-screen SI-SNR that much, but does result in a OSR reduction of at least 0.6 dB for single mixtures and 1.7 dB for MoMs.
Thus, attentional pooling seems to have a beneﬁcial effect in that it improves off-screen suppression, with equivalent classiﬁcation and on-screen separation performance.
A.3. D ATA FILTERING
As described in section 5.
[Table]
we use an unsupervised audio-visual coincidence model (Jansen et al., 2020) to ﬁlter training videos for on-screen sounds.
To ablate the beneﬁt of this ﬁltering
[Table]
we tried using different combinations of ﬁltered and unﬁltered data for NOn examples, as described in section 5.2, which uses ﬁlterd data for both on-screen and off-screen mixtures.
Filtered data has the advantage of less noisy on-screen labels
[Table]
but the disadvantage that it lacks the variety of unﬁltered data, being only 4.7% of the unﬁltered data.
[Table]
The results are shown in Table 6.
For unsupervised training, unﬁltered on-screen with ﬁltered off-screen achieves improved performance in terms of AUC-ROC and on-screen SI-SNR, yet OSR decreases for MoMs.
This suggests that in the absence of cleanly-labeled on-screen videos, a larger amount of data with noisier labels is better compares to a smaller amount of data with less noisy labels.
However, for semi-supervised training that includes a small amount of cleanly-labeled on-screen examples, AUC-ROC is consistently worse for all ablations, and on-screen SI-SNR and OSR are generally equivalent or worse for all ablations.
Thus, these ablations validate that using ﬁltered data for both on-screen and off-screen components of NOn examples with semi-supervised training achieves the best results overall.
A.3. N UMBER OF OUTPUT SOURCES
For all experiments in this paper
[Table]
we generally used M = 4 output sources for the separation model, which is the maximum number of sources the model can predict.
Here we see if increasing the number of output sources can improve performance.
More output source slots provides a separation model with more ﬂexibility in decomposing the input waveform, yet the drawback is that the model may over-separate (i.e. split sources into multiple components), and there is more pressure on the classiﬁer to correctly group components of the on-screen sound together.
The results are shown in
Table 7.
[Table]
For unsupervised training, increasing the number of output sources generally degrades AUC-ROC and on-screen SI-SNR
[Table]
while boosting OSR a bit.
Note that the MixIT* improves for MoMs with 8 output sources (10.5 dB → 11.1 dB)
[Table]
which suggests the greater ﬂexibility of the model, yet the on-screen estimate ˆ x on is quite a bit worse (3.6 dB), also compared to on-screen SI-SNR for 4 output sources (6.3 dB).
For semi-supervised training, MixIT* performance also improves with more output sources, but
AUC-ROC and on-screen SI-SNR decrease, suggesting the increased pressure on the classiﬁer to make correct predictions for more, and potentially partial, sources.
OSR increases with more output sources, which suggests the classiﬁer biases towards predicting 0s more often.
Thus, increasing the number of sources shifts the operating point of the model away from separating on-screen sounds and towards suppressing off-screen sounds.
A.3. B ASELINE SEPARATION MODEL
We also trained two-output baseline separation models without the on-screen classiﬁer, where the ﬁrst estimated source is the on-screen estimate ˆ x on with training target of on-screen audio, and the second estimated source is the off-screen estimate ˆ x oﬀ with training target of off-screen audio.
These models were trained with or without video conditioning, using the negative SNR loss (3).
The training data is exactly the same as in Table 1, with 0% SOff.
Table 8 shows the results in terms of the same metrics used in Tables 1, except that instead of “off-screen rejection ratio (OSR)”, we report “input-to-source ratio (ISR)”
(i.e. 10 log 10 of the ratio of input power to estimated source power) for each of the two output sources.
High ISR means that the source power is lower compared to the input power.
Note that ISR ˆ x on for off-screen single-mixtures and MoMs is equivalent to OSR.
Table 8 also includes several trivial baselines with expected scores.
First, notice that none of these models approach the performance of separation models that include the on-screen classiﬁer, as shown in Table 1.
Second, the unsupervised and semi-supervised models here achieve distinctly different operating points.
Without video conditioning, the unsupervised model achieves a trivial solution, nearly equivalent to just outputting 1 / 2 the input mixture for each estimated source.
Adding video conditioning for the unsupervised model actually reduces single-mixture performance a bit (66.2 dB to 29.7 dB).
The semi-supervised model without video conditioning is very poor at single-mixture on-screen
SI-SNR (-18.0 dB), yet achieves quite high single-mixture OSR (around 50 dB).
As indicated by the ISRs, the model tends to prefer nearly-zero on-screen estimates, which may be due to the additional cleanly-labeled off-screen examples provided during training.
For the video-conditioned semi-supervised model, single-mixture on-screen SI-SNR improves by quite a lot (-18.0 dB to 18.
[Table]
dB), but
on-screen SI-SNR performance for on-screen MoMs is abysmal (-19.7 dB without visual conditioning, -5.3 dB with visual conditioning).
Overall, we can conclude from these baselines that simply training a two-output separation model with on-screen and off-screen targets, even with visual conditioning, is not a feasible approach for our open-domain and noisily-labeled data.
A. N EURAL NETWORK ARCHITECTURES
We brieﬂy present the architectures used in this work for the separation network M s , the audio embedding network M a , and the image embedding network M v , and referred in Sections 3.1, 3.
and 3.3, respectively.
We present the architecture of the TDCN++ separation network in Table 9.
The input to the separation network is a mixture waveform with T time samples and the output is a tensor containing the
M estimated source waveforms ˆ s ∈ R M × T .
The input for the i th depth-wise (DW) separable convolutional block is the summation of all skip-residual connections and the output of the previous block.
Speciﬁcally, there are the following skip connections deﬁned w.r.t.
their index i = 0 , . .
. , 31 : 0 (cid:1) 8 , 0 (cid:1) 16 , 0 (cid:1) 24 , 8 (cid:1) 16 , 8 (cid:1) 24 and 16 (cid:1) 24 .
For more information we refer the reader to a detailed description of the model in Wisdom et al.
(2020b).
In a similar way, in Table 10 we deﬁne the image and audio embedding networks, which use the same
MobileNet v1 architecture (Howard et al., 2017) with different input tensors.
The extraction of each image embedding Z v j , j = 1 , . .
. , 5 relies on the application of the image embedding network M v on top of each input video frame individually.
Moreover, in order to extract the local video spatio-temporal embedding, we extract the output of the 8 × 8 convolutional map (denoted with a * in Table 10) for each input video frame and feed it through a dense layer in order to reduce its channel dimensions to 1 .
By concatenating all these intermediate convolutional maps we form the local spatio-temporal video embedding Z vl as speciﬁced in Section 3.3.
On the other hand, we extract a time-varying embedding Z a m for the m th separated source waveform by applying the audio embedding network M a on overlapping audio segments and concatenating those outputs.
The audio segments are extracted with an overlap of 86 windows or equivalently 0 . 86 seconds. Speciﬁcally, for each segment, we extract the mel-spectrogram representation fromwindows with a length of 25 ms and a hop size of 10 ms forming the input for the audio embedding network as a matrix with size 96 × 64 , where 64 is the number of mel-features. After feeding this
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