cometadata/datacite-titles-descriptions-related-identifiers huggingface dataset:
DOI: 10.35111/2n7h-n869
title: CSLU: Speaker Recognition Version 1.1

description: <h3>Introduction</h3> <p>This file contains documentation on the CSLU Speaker Recognition Corpus, Version 1.1, Linguistic Data Consortium (LDC) catalog number LDC2006S26 and ISBN 1-58563-382-8. </p><p>The Speaker Recognition corpus (formerly known as Speaker Verification), consists of telephone speech from 91 participants. Each participant has recorded speech in twelve sessions over a two-year period answering questions like "what is your eye color" or responding to prompts like "describe a typical day in your life." Most of the utterances in the release of the corpus have corresponding non-time-aligned word level transcriptions. </p><p> In most of the CSLU data collections, each participant calls a toll free telephone number and answers a few question. CSLU records the speech, transcribes it, then packages it as a released corpus. </p><p> The Speaker Recognition data collection was quite a bit more complicated. The goal of the data collection was to collect speech from each participant over a two-year period. Each participant called call the data collection system 12 times over the two-year period and say the same utterances each time. </p><p> Some of the recording sessions were only a few days apart and others several weeks apart. Participant followed the following calling schedule. During the first month, they called twice in a week. No calls were made in the second and third months. In the fourth month they made one call. No calls were made in the fifth and sixth months. This pattern repeated three more times for a total of 12 calls per participant.</p><p> In order to balance the workload required to remind participants to call and to avoid large data collection bursts on the system, the participants were divided into 12 groups. Each group began the two-year schedule on subsequent months. The first group started in September 1996. The second group started in October 1996. And so on. </p><h3>Samples</h3> <p>For an example of the data in this corpus, please listen to the following audio <a href="./desc/addenda/LDC2006S26.wav" rel="nofollow">sample</a>. </p> </br>
Portions © 1996-2002 Center for Spoken Language Understanding, Oregon Health & Science University, © 2006 Trustees of the University of Pennsylvania
DOI: 10.35111/3bpn-2713
title: Levantine Arabic QT Training Data Set 5, Speech

description: <h3>Introduction</h3><br>
<p>Levantine Arabic QT Training Data Set 5, Speech contains 1,660 calls totalling approximately 250 hours of telephone conversation in Levantine Arabic. These calls were collected between 2003 and 2005. Corresponding transcriptions may be found in <a href="../../../LDC2006T07">LDC2006T07</a>.</p><br>
<h3>Data</h3><br>
<p>This corpus is the combination of four former training data sets: LDC2004E21 and LDC2004E22, LDC2004E65 and LDC2004E66, LDC2005S07 and LDC2005T03 and LDC2005S14 (Speech and Transcripts). More than half of the speakers are Lebanese, the others are Jordanian, Palestinian, and Syrian. The table below shows the distribution of the speakers' national origin:</p><br>
<ul><br>
<li>559 Jordanian</li><br>
<li>1,853 Lebanese</li><br>
<li>355 Palestinian</li><br>
<li>67 Syrian</li><br>
<li>484 Levantine speakers whose national origin could not be determined.</li><br>
</ul><br>
<h3>Samples</h3><br>
<p>For an example of this corpus, please listen to this <a href="desc/addenda/LDC2006S29.wav" rel="nofollow">audio sample</a> in sphere format.</p><br>
<h3>Updates</h3><br>
<p>Eight sphere files were found to have incorrect headers. As of October, 2017 these files have been fixed. All copies downloaded after this date will have the correct files.</p></br>
Portions © 2003-2006 Trustees of the University of Pennsylvania
DOI: 10.35111/6kce-vx52
title: Speech Controlled Computing

description: <h3>Introduction</h3><br>
<p>This file contains documentation on Speech Controlled Computing, Linguistic Data Consortium (LDC) catalog number LDC2006S30 and ISBN 1-58563-380-1.</p><br>
<p>The Speech Controlled Computing corpus was designed to support the development of small footprint, embedded ASR applications in the domain of voice control for the home. It consists of the recordings of 125 speakers of American English from four dialect regions, three age groups and two gender groups, pronouncing isolated words. The four primary dialect regions covered by the corpus are North, South, West and Midland as defined by Williams Labov's Atlas of North American English. The three primary age groups covered by the corpus are 18-29, 30-49 and 50+.</p><br>
<p>The recordings were conducted in a sound-attenuated room at LDC with the AKG C4000B studio condenser microphone. The omni-directional mode of the C4000B was used. Each speaker read a randomized word list consisting of 2,100 words (100 distinct words appearing 21 times each). Speech utterances were digitized and recorded to a DAT, as well as to a hard disk drive via the Townshend DATLINK+ digital audio interface.</p><br>
<p>Speech utterances were audited as they were recorded, and any utterances detected by the recorder that were not spoken clearly or correctly were re-recorded. This included extraneous clicks, coughs, sighs and breathing that may have corrupted the recorded words. Utterances that were spoken too soft or too loud were also re-recorded.</p><br>
<p>The digitized utterances were automatically segmented and aligned to the word list. Then each utterance was audited and the segmentation was checked, and corrected if necessary, by an annotator using an auditing and segmenting tool developed by LDC.</p><br>
<p>Finally, sound files containing individual utterances were generated using the alignment and segmentation information. The sound files for this corpus were created with 100 msec of silent time before and after each utterance. Any files that contained noticeable clipping were automatically removed.</p><br>
<h3>Samples</h3><br>
<p>For an example of this corpus, please listen to this audio <a href="desc/addenda/LDC2006S30.wav" rel="nofollow">sample</a></p></br>
 © 2003-2006 Trustees of the University of Pennsylvania
DOI: 10.35111/38cj-3k75
title: 2003 NIST Language Recognition Evaluation

description: <h3>Introduction</h3><br>
<p>The goal of the <a href="https://www.nist.gov/itl/iad/mig/language-recognition">NIST Language Recognition Evaluation (LRE)</a> is to establish the baseline of current performance capability for language recognition of conversational telephone speech and to lay the groundwork for further research efforts in the field. The series had its first evaluation in 1996. 2003 NIST Language Recognition Evaluation (LRE-03) was part of this ongoing series of evaluations of language recognition technology.</p><br>
<p>The task evaluated was the detection of a given target language. Languages included in this releasea are Arabic, English, Farsi, French, German, Hindi, Japanese, Korean, Mandarin, Russian, Spanish, Tamil and Vietnamese.<br />Given a test segment of speech, a target language was assigned as a test hypothesis, and the task was to determine whether this test hypothesis was true or false. This release contains both the 1996 and 2003 NIST Language Recognition Evaluations.</p><br>
<p>LDC released other LREs as:</p><br>
<ul><br>
<li>2005 NIST Language Recognition Evaluation (<a href="../../../LDC2008S05">LDC2008S05</a>)</li><br>
<li>2007 NIST Language Recognition Evaluation Test Set (<a href="../../../LDC2009S04">LDC2009S04</a>)</li><br>
<li>2007 NIST Language Recognition Evaluation Supplemental Training Set (<a href="../../../LDC2009S05">LDC2009S05</a>)</li><br>
<li>2009 NIST Language Recognition Evaluation Test Set (<a href="../../../LDC2014S06">LDC2014S06</a>)</li><br>
<li>2011 NIST Language Recognition Evaluation Test Set (<a href="../../../LDC2018S06">LDC2018S06</a>)</li><br>
</ul><br>
<h3>Data</h3><br>
<p>Each speech file is one side of a "four wire" telephone conversation represented as 8-bit, 8kHz mulaw data. There are 11,830 speech files in sphere(.sph) format for a total of around forty six hours of speech. The speech data was compiled from the LDC's CALLFRIEND, CALLHOME, and Switchboard-2 corpora. Each file contains one test segment. The test segments are divided into three-second, ten-second, and thirty-second tests, each in its own directory.</p><br>
<h3>Samples</h3><br>
<p>For an example of the data in this corpus, please listen to this audio <a href="desc/addenda/LDC2006S31.wav" rel="nofollow">sample</a>.</p><br>
<h3>Updates</h3><br>
<p>A typo was fixed in the index.html file. There are 11,830 sphere files, not 11,839. The updated index file is available in the <a href="docs/LDC2006S31/" rel="nofollow">online docs</a> folder.</p></br>
Portions © 1996-2002, 2006 Trustees of the University of Pennsylvania
DOI: 10.35111/sk8b-ss58
title: Middle East Technical University Turkish Microphone Speech v 1.0

description: <h3>Introduction</h3><br>
<p>Middle East Technical University Turkish Microphone Speech v 1.0 was developed at <a href="http://www.metu.edu.tr/">Middle East Technical University (METU)</a> as part of a collaborative work between METU's Department of Electrical and Electronics Engineering and the Center for Spoken Language Research (CSLR) at the University of Colorado at Boulder. The collaboration was supported by TUBITAK, the Scientific and Technical Research Council of Turkey, through a combined doctoral scholarship program. The corpus was used to port CSLR's speech recognition system, SONIC, to Turkish.</p><br>
<p>The corpus contains text, speech and alignment files. The corpus is of size ~600Mbytes. 120 speakers (60 male and 60 female) speak 40 sentences each (aproximately 300 words per speaker), which makes approximately 500 minutes of speech in total. The 40 sentences are selected randomly for each speaker from a triphone-balanced set of 2,462 Turkish sentences. The speakers are selected from students, faculty and staff at METU and all are native speakers of Turkish. The age range is from 19 to 50 years with an average of 23.9 years.</p><br>
<p>The data has been digitally recorded with a Sound Blaster sound card on a PC at a 16 kHz sampling rate.</p><br>
<h3>Samples</h3><br>
<p>Please listen to this <a href="desc/addenda/LDC2006S33.wav" rel="nofollow">audio sample</a> and examine its companion <a href="desc/addenda/LDC2006S33.txt" rel="nofollow">transcript</a> for an example of the data contained in this publication.</p></br>
Portions © 2001, 2002, 2005 Middle East Technical University, Tolga Ciloglu, Ozgul Salor, Bryan Pellom, Kadri Hacioglu, Mubeccel Demirekler, © 1993, 2006 Trustees of the University of Pennsylvania
DOI: 10.35111/bw5g-8741
title: Russian through Switched Telephone Network (RuSTeN)

description: <h3>Introduction</h3><br>
<p>Russian through Switched Telephone Network (RuSTeN) was developed in the Automatic Voice Identification System in Telephone Channel project. The purpose of the project was to develop software for automatic identification of speakers based on voice samples acquired through telephone channels. System training was performed with the RuSTeN corpus.</p><br>
<h3>Data</h3><br>
<p>The RuSTeN database was recorded between March 2001 and February 2003 by Speech Technology Center (STC) using the "forget-me-not" professional telephone recording and archiving software package developed by STC. The files were recorded with sample frequency 11025 Hz, one-channel, 16-bit linear.</p><br>
<p>Each of the speakers made at least five calls from different locations and/or telephone sets. Most of the calls were made from home or an office environment with uncontrolled noise level. Additionally, one call per speaker was made from a public telephone (with either street or metro station noise in the background). The recordings are spontaneous (sometimes guided by the near-end speaker) conversations between the caller and the speech database collector on various subjects (the weather, the caller's biography, hobbies, etc.) and include approximately 150 seconds of the far-end and at least five seconds of the near-end speaker. Besides, each time the caller was asked to utter the usual digits set (0-9) and the words "yes" and "no."</p><br>
<p>The time interval between two successive sessions is at least two days. The database contains 125 speakers (far-end), 58 male and 67 female. Each far-end speaker is represented by at least five speech files. The sound files are in the wav-format. The speech filenames contain the following information: FFF (far-end speaker number) and SS (session number).</p><br>
<h3>Samples</h3><br>
<p>For an example of the data in this corpus, please review this <a href="desc/addenda/LDC2006S34.wav" rel="nofollow">audio sample</a>.</p></br>
Portions © 2001 Speech Technology Center Limited, © 2006 Trustees of the University of Pennsylvania
DOI: 10.35111/j0p6-f049
title: CSLU: Multilanguage Telephone Speech Version 1.2

description: <h3>Introduction</h3> <p>The Multilanguage Telephone Speech corpus consists of telephone speech from 11 languages: English, Farsi, French, German, Hindi, Japanese, Korean, Mandarin, Spanish, Tamil, Vietnamese. The corpus contains fixed vocabulary utterances (eg. days of the week) as well as fluent continuous speech. The current release includes recorded utterances from about 2,052 speakers, for a total of about 38.5 hours of speech. Time-aligned phonetic transcriptions for 619 of the utterances are also included. </p><h3>Data</h3> <p>Each subject called the CSLU data collection system by dialing a toll-free number. An analog telephone line was connected to a Gradient Technologies box. Data from incoming calls were recorded by the Gradient box. The sampling rate was 8 khz and the files were stored in 16-bit linear format on a UNIX file system. Each utterance was recorded as a separate file.</p><h3>Samples</h3> <p>For an example of the data in this corpus, please listen to these audio samples in <a href="./desc/addenda/LDC2006S35_Tam.wav" rel="nofollow">Tamil</a> and <a href="./desc/addenda/LDC2006S35_Eng.wav" rel="nofollow">English</a>. </p></br>
Portions © 1992, 2000, 2002 Center for Spoken Language Understanding, Oregon Health & Science University, © 2006 Trustees of the University of Pennsylvania
DOI: 10.35111/ezr5-1t33
title: West Point Korean Speech

description: <h3>Introduction</h3> <p>This file contains documentation on West Point Korean Speech, Linguistic Data Consortium (LDC) catalog number LDC2006S36 and ISBN 1-58563-360-7. </p><p>West Point Korean Speech is a database of digital recordings of spoken Korean. Corpus design and data collection were carried out by staff and faculty of the Department of Foreign Languages (DFL) and Center for Technology Enhanced Language Learning (CTELL), located at the United States Military Academy (USMA), West Point, New York. The corpus was designed to develop speech recognition systems that would be used by the US government for speech-recognition enhanced language learning courseware. The prompt scripts were created from 20,000 distinct sentences, along with a subset of prompts designed to elicit free response answers to questions for use in domain-specific speech to speech translation systems. Each speaker attempted to record 100 utterances. Three data collection scripts were designed by Ms. Jennifer Son, a native speaker of Korean, under contract with the Department of Foreign Languages.</p> <h3>Samples</h3> <p>For an example of the data in this corpus, please listen to the following <a href="./desc/addenda/LDC2006S36.wav" rel="nofollow">audio sample</a>. </p> </br>
Portions © 2002 United States Military Academy, Portions © 2006 Trustees of the University of Pennsylvania.
DOI: 10.35111/qyw6-w652
title: CSLU: Names Release 1.3

description: <p> A common problem in training and developing speech recognition systems is scarcity of data, especially particular phonemic contexts. The Center for Spoken Language Understanding is attempting to address this problem with the Names Corpus. The Names Corpus is a collection of name utterances, both first and last names, from several thousand different speakers over the telephone. Name utterances are "spontaneous" in that the subject is not reading from a word list. </p><p> Another area of active research is the development of name Recognition systems. The Names Corpus is a useful resource for addressing this problem. </p> <p> The utterances in this corpus were taken from many other telephone speech data collections that have been completed at the CSLU. In most data collections, the callers were asked to leave their name at some point. Also, the callers would occasionally leave their name in the midst of another utterance. The names in these situations were extracted out of the host utterance and added to the Names Corpus. </p> <p> Each file in the Names Corpus has an orthographic transcription following the CSLU Labeling Conventions. Also, to take advantage of the phonemic variability, many of the utterances have been phonetically transcribed. The selection of files to phonetically transcribe was constrained by a process that selected files that were suspected to contain phonetic contexts that had not yet been transcribed. </p> <p> Release 1.3 of this corpus contains 24,245 files. All of these have been phonetically labeled. Approximately 40% of the bigram phonemic contexts possible, without regard to language constraints, are represented. </p> <h3>Samples</h3> <p> For an example of the data in this publication, please review this audio <a href="./desc/addenda/LDC2006S39.wav" rel="nofollow">sample</a> and its <a href="./desc/addenda/LDC2006S39.txt" rel="nofollow">transcription</a>. </p>  </br>
Portions © 2001, 2003 Speech Technology Center Ltd., © 2006 Trustees of the University of Pennsylvania
DOI: 10.35111/edfk-fe78
title: Korean Broadcast News Speech

description: <h3>Introduction</h3> <p>This data set consists of 18 audio files recorded by LDC in January 2000 and February 2000 from Voice of America (VOA) satellite radio news broadcasts in Korean. </p><h3>Data</h3> <p>The recordings, captured from a dedicated satellite receiver, are stored as 16-bit PCM, 16-kHz, single-channel, in NIST SPHERE format. The duration of each recording is either 30 minutes or 60 minutes, depending on the VOA broadcast schedule. The date (YYYYMMDD), start-time and end-time (HHMM, Eastern Standard Time) for each recording are indicated in its file name. The sample data is not compressed. </p><p>Transcripts for these recordings are available as a separate corpus from the LDC: Korean Broadcast News Transcripts, LDC2006T14. </p><h3>Samples</h3> <p>For an example of the data contained in this corpus, please listen to this <a href="./desc/addenda/LDC2006S42.wav" rel="nofollow">audio sample</a> (wav format). </p> </br>
Portions © 2000, 2006 Trustees of the University of Pennsylvania
DOI: 10.35111/nsvg-dd69
title: Gulf Arabic Conversational Telephone Speech

description: <h3>Introduction</h3> <p> This database contains 975 Gulf Arabic speakers taking part in spontaneous telephone conversations in Colloquial Gulf Arabic. A total of 976 conversation sides are provided (one speaker appears on two distinct calls). The average duration per side is about 5.7 minutes.</p> <p>This corpus was collected and transcribed in 2004 by Appen Pty Ltd (Appen), Sydney, Australia.</p> <h3>Data</h3> <p>The single-channel files represent just one side of a normal conversation. The "devtest" set represents a relatively balanced (representative) sample drawn from the total pool of collected calls, based on a test-set selection process applied by the National Institute of Standards and Technology (NIST) and based on demographic, phone and audit information as provided by Appen.</p> <h3>Samples</h3> <p> For an example of the data contained in this corpus, please listen to this <a href="./desc/addenda/LDC2006S43.wav" rel="nofollow">audio sample(wav)</a>. </p> </br>
Portions © 2006 Trustees of the University of Pennsylvania
DOI: 10.35111/rawd-1051
title: 2004 NIST Speaker Recognition Evaluation

description: <h3>Introduction</h3><br>
<p>2004 NIST Speaker Recognition Evaluation consists of English conversational telephone speech collected by the Linguistic Data Consortium and corresponding transcripts used in the 2004 evaluation.</p><br>
<p>This release is part of an ongoing series of yearly evaluations conducted by NIST (National Institute of Standards and Technology). These evaluations provide an important contribution to the direction of research efforts and the calibration of technical capabilities. They are intended to be of interest to all researchers working on the general problem of text-independent speaker recognition. To this end the evaluation was designed to be simple, to focus on core technology issues, to be fully supported, and to be accessible.</p><br>
<p>NIST has been coordinating Speaker Recognition Evaluations since 1996. Each evaluation begins with the announcement of the official evaluation plan which clearly states the rules and tasks involved with the evaluation. The evaluation culminates with a follow-up workshop, where NIST reports the official results and researchers share in their findings.</p><br>
<p>Additional documentation is available in the <a href="https://www.nist.gov/sites/default/files/documents/2017/09/26/sre-04_evalplan-v1a.pdf">NIST 2004 SRE Evaluation Plan </a>.</p><br>
<h3>Samples</h3><br>
<p>This <a href="desc/addenda/LDC2006S44.wav" rel="nofollow">audio sample</a> and its <a href="desc/addenda/LDC2006S44_trans.txt" rel="nofollow">transcript</a> provide an example of the data contained in this corpus.</p></br>
Portions © 2004, 2006 Trustees of the University of Pennsylvania
DOI: 10.35111/2dcs-9751
title: Iraqi Arabic Conversational Telephone Speech

description: <h3>Introduction</h3> <p>This database contains 474 Iraqi Arabic speakers taking part in spontaneous telephone conversations in Colloquial Iraqi Arabic. A total of 478 conversation sides are provided (most speakers appear only once), and most of these call sides comprise both sides of a conversation (that is, 202 two-channel recordings plus 74 single-channel recordings). The average duration per call is about 6 minutes, so each call side contains about 3 minutes of speech, on average.</p> <p>This corpus was collected and transcribed in 2003 and 2004 by Appen Pty Ltd, Sydney, Australia.</p> <h3>Samples</h3> <p>For an example of the speech contained in this corpus, please list to this <a href="./desc/addenda/LDC2006S45.wav" rel="nofollow">sample audio file</a> in wav format. </p> </br>
 Portions © 2006 Trustees of the University of Pennsylvania