SSEmb: Training a Generalizable Speech Retrieval Model Using Only Automatic Speech Recognition Data
Authors: Sangkwon Park, Dongwook Lee, Youngho Cho, Donghun Kang, Heeseung Kim, Sungroh Yoon
Organizations: ECE, Seoul National University · IPAI, Seoul National University · University of Seoul · AIIS, ASRI, INMC, and ISRC, Seoul National University
End-to-end speech-to-text retrieval is becoming an important capability for voice agents, allowing spoken queries to retrieve relevant documents without intermediate transcription. Training these retrievers typically depends on large collections of spoken query-document pairs that are costly to acquire and difficult to extend across domains and languages. We introduce \ours, which replaces such task-specific supervision with speech-transcript pairs from standard ASR corpora. A lightweight adapter projects the output of a frozen speech encoder into the retrieval space of a frozen pretrained text embedding model. \ours is competitive with cascaded ASR-then-retrieve systems while reducing inference latency, and outperforms existing publicly available speech embedding models. It also transfers to text-to-speech and speech-to-speech retrieval without additional training. Controlled analyses validate the speech-transcript alignment paradigm, demonstrating robustness to reduced training data and consistent gains from larger corpora. Further ablations of loss functions and trainable modules establish practical guidelines for the framework. Model weights and codes are to be released.
Figures & tables
Figure 1: Overview of the SSEmb architecture. The acoustic representation of speech passes through an adapter into the text embedding model to produce speech embeddings. ASR data is used to align the embeddings of speech with those of its transcript. At inference, the speech query is embedded in the same manner for retrieval.
HotpotQA
MuSiQue
Mr.TyDi (Eng)
Model
R@2
R@5
nDCG@10
R@2
R@5
nDCG@10
R@2
R@5
nDCG@10
Δt (speedup)
Cascaded Q3E-0.6B
0.4940
0.6942
0.7267
0.4535
0.6614
0.7201
0.6453
0.8356
0.6976
0.348
SSEmb (Q3E-0.6B)
0.4736
0.6729
0.7049
0.4162
0.6141
0.6764
0.5703
0.7860
0.6450
0.144 ( 2.42 × )
Cascaded Q3E-4B
0.5369
0.7553
0.7772
0.4839
0.6987
0.7494
0.7048
0.8774
0.7429
0.386
SSEmb (Q3E-4B)
0.5215
0.7373
0.7609
0.4353
0.6484
0.6998
0.6659
0.8495
0.7135
0.145 ( 2.66 × )
Cascaded Q3E-8B
0.5425
0.7656
0.7886
0.5068
0.7277
0.7788
0.7389
0.8875
0.7616
0.405
Table 1: Retrieval performance across three English benchmarks. Shaded rows indicate cascaded systems corresponding to each model, where the transcript of the input query is directly used for retrieval. Δt denotes the latency of each method, measured in seconds. Speedup over the corresponding cascaded baseline is also reported.
Korean (960h)
Japanese (315h)
Thai (40h)
Model
R@2
R@5
nDCG@10
R@2
R@5
nDCG@10
R@2
R@5
nDCG@10
Cascaded
0.6715
0.8265
0.7097
0.6430
0.8380
0.6816
0.6387
0.8037
0.6934
SSEmb
0.5805
0.7805
0.6568
0.5210
0.7310
0.5864
0.6615
0.8468
0.7087
SONAR
0.1770
0.3470
0.2975
0.1360
0.3090
0.2607
0.1403
0.3018
0.2567
Indonesian (35h)
Russian (55h)
Arabic (80h)
Model
R@2
R@5
nDCG@10
R@2
R@5
nDCG@10
R@2
R@5
nDCG@10
Table 2: Retrieval performance of SSEmb on Mr.TyDi across six low-resource languages.
R@1
R@5
R@10
nDCG@10
Text2Speech
Cascaded
0.6978
0.9086
0.9507
0.8276
SSEmb
0.6997
0.9168
0.9632
0.8342
WavRAG †
0.6844
0.8374
0.9023
0.8483
Speech2Speech
Cascaded
0.6870
0.9047
0.9497
0.8218
Table 3: Retrieval performance on SpokenSquad comparison across Text2Speech and Speech2Speech settings. WavRAG † results are taken directly from the original paper [ 3 ] .
HotpotQA
MuSiQue
R@2
R@5
nDCG
R@2
R@5
nDCG
Cascaded
0.4940
0.6942
0.7267
0.4535
0.6614
0.7201
Ours
0.4736
0.6729
0.7049
0.4162
0.6141
0.6764
Training paradigm
w/ ASR
0.4936
0.6943
0.7240
0.4253
0.6286
0.6921
w/ Q-D
0.4593
0.6627
0.6879
0.3796
0.5915
0.6475
Table 4: Ablation study on training strategies using the 0.6B model. We compare different training paradigms, the amount of training data, loss, and trained modules. Cascaded denotes the upper-bound pipeline. nDCG refers to nDCG@10.
Existing speech retrieval systems rely on fixed similarity matching and cannot adapt to diverse user intents. We introduce INSPIRE, the first benchmark for instruction-aware speech retrieval, in which natural-language instructions dynamically specify relevance criteria, including semantic content, speaker identity, speaking style, environmental sounds, and their combinations. We evaluate four retrieval paradigms: large audio-language models, cascaded pipelines, self-supervised speech models, and contrastive audio-language models. Our results reveal that no current method robustly handles all retrieval intents. Text-based approaches perform relatively better at semantic retrieval but struggle with paralinguistic attributes, while speech-based models are moderately better at capturing acoustic properties but falter at following instructions. These findings highlight the need for unified architectures capable of instruction-aware speech retrieval.
Chen-An Li, Hung-yi Lee
National Taiwan University · NTU Artificial Intelligence Center of Research Excellence (NTU AI-CoRE)
Speech retrieval has become increasingly important as spoken content continues to grow across meetings, lectures, podcasts, and videos. Existing benchmarks and models have advanced semantic search over spoken content, but largely focus on what is said while overlooking who says it. In many real-world scenarios, however, users need to retrieve speech based jointly on semantic content and a target speaker, where the speaker may be specified naturally through a reference speech utterance rather than a predefined identity. To address this gap, we introduce VoiceTrace-Bench, a benchmark for hybrid speech retrieval in which each query combines text specifying what to retrieve with reference speech specifying who to retrieve. This setting requires models to integrate complementary semantic and speaker information directly from heterogeneous query inputs. Motivated by the joint audio-text modeling capabilities of audio-language models (ALMs), we develop VoiceTrace, a two-stage retrieval framework consisting of VoiceTrace-Emb, an embedding model that learns unified representations for efficient large-scale retrieval, and VoiceTrace-Reranker, a reranking model that jointly examines each query--candidate pair for fine-grained relevance estimation. Experiments show that VoiceTrace achieves state-of-the-art performance on established semantic speech retrieval benchmarks, while substantially outperforming cascade-based approaches on VoiceTrace-Bench, demonstrating its effectiveness for both conventional semantic retrieval and the new hybrid retrieval setting.
Aaron Yee, Fengjie Lu, Jiarui Hai +5
Zhejiang University · Johns Hopkins University · independent researcher
Speech LLM-based ASR often struggles with named entities and long-tail words due to strong internal language-model priors. Retrieval-augmented biasing can help, but its effectiveness depends on accurate hotword localization in full-utterance speech under weak supervision. We propose CLAR, a dual-encoder speech-text retriever that uses Continuous Integrate-and-Fire (CIF) to learn monotonic token-level alignments without timestamps. With length-aware localized matching, CLAR anchors short-entity acoustic cues and reduces representation dilution and attention drift. The retriever is trained with a multi-granularity objective combining global and local segment-level contrastive losses and a CIF quantity constraint. At inference, top-ranked hotwords are injected as contextual prompts for the Speech LLM, improving recognition without shallow fusion. Experiments show that CLAR significantly improves hotword retrieval and reduces both CER and B-WER against strong contextual ASR baselines.