Setting Up an Audio Database for Music Information Retrieval Benchmarking
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英语复习第一单元A篇:1.Research suggests that our planet is warming at a(n) pace.1. 研究表明,我们的星球正在以前所未有的速度变暖。
2. Your rate is the speed at which your body transforms food into energy.2. 新陈代谢率是指身体将食物转化为能量的速度。
3. The research team proposes a few key recommendations to the declining of domestic demand.3. 该研究小组提出了一些应对国内需求下降的关键建议。
4. A(n) management system optimizes battery range in high and low temperatures, allowing the car to maintain optimal range capability.4. 热管理系统可优化电池在高温和低温下的续航能力,使汽车保持最佳的续航能力。
5. By the 1920s, from winds and weather had removed a large part of the land.5。
到20世纪20年代,由于风和天气的侵蚀,大部分土地已经消失。
6.It's needless to say that the value of water, this and irreplaceable resource, is enormous.6. 毋庸置疑,水这种有限而不可替代的资源的价值是巨大的。
7. Experts say the spread of the virus in countries with little protection from vaccines could lead to more variants and the pandemic.7. 专家表示,病毒在缺乏疫苗保护的国家传播可能会导致更多变异,并延长大流行的时间。
深度学习在音频识别中的应用音频识别是指根据声音信号来识别音频内容,这在语音识别、音乐识别等领域都有着广泛的应用。
在过去的几十年中,随着人工智能技术的不断发展,深度学习作为一种强大的机器学习方法在音频识别中得到了广泛应用。
深度学习模型可以通过训练大量的音频数据,学习到音频信号的特征表示,从而实现对音频内容的准确识别和分类。
一、深度学习在音频特征提取中的应用主要集中在音频特征提取方面。
传统的音频处理方法通常是基于手工设计的特征提取算法,如MFCC(Mel频率倒谱系数)、频谱特征等。
这些传统方法对于不同类型的音频数据效果不同,且需要大量的人力和时间来进行特征工程。
而深度学习的特点是可以学习到数据中的高阶特征表示,从而减少了对手工设计特征的需求,同时能够更好地适应不同类型的音频数据。
在音频识别任务中,深度学习模型通常以卷积神经网络(CNN)或循环神经网络(RNN)为基础。
CNN可以有效地捕捉音频数据中的时空信息,对于音频信号的局部特征有较好的提取能力;而RNN则适用于处理时序信息,可以帮助模型更好地理解音频数据中的时间关系。
另外,还有一种常用的深度学习结构是深度神经网络(DNN),它可以通过多层的非线性变换来学习数据中的高阶特征表示,从而提高音频识别的准确率。
二、深度学习在语音识别中的应用语音识别是音频识别的一个重要应用场景,深度学习在语音识别领域也有着广泛的应用。
传统的语音识别系统通常基于高斯混合模型(GMM)和隐马尔可夫模型(HMM)构建的系统,需要大量的人工调参和特征工程来提高准确率。
而深度学习在语音识别中的应用,主要是通过深度神经网络(DNN)和循环神经网络(RNN)等模型来实现端到端的语音识别。
近年来,随着深度学习技术的不断发展,基于深度学习的语音识别系统取得了很大的进展。
其中,长短时记忆网络(LSTM)和注意力机制(Attention)在语音识别任务中得到了广泛的应用。
LSTM网络可以有效地处理长序列数据,保留了序列数据中的重要信息;而注意力机制可以帮助模型更好地关注输入序列中的重要部分,提高了语音识别的准确率。
audiotrack 用法[audiotrack 用法],以中括号内的内容为主题,写一篇1500-2000字文章,一步一步回答一、什么是audiotrack?audiotrack 是一个Android 平台上的类,用于管理和控制音频数据播放。
按照官方文档的说法,AudioTrack 是" 用于播放或录制音频的类,可直接访问和处理音频数据流。
如果需要播放流式音频,建议使用MediaPlayer 类"(译文)。
二、如何使用audiotrack?1. 创建一个audiotrack 对象要使用audiotrack,首先需要创建一个audiotrack 对象。
创建audiotrack 时,需要设置一些参数,包括:- 音频切片的数据属性,如采样率、采样位数、声道数等;- 播放模式,包括播放的数据格式和连接模式;- 写入模式,是静态写入模式还是流式写入模式;- 播放缓冲区的大小。
我们可以使用以下代码创建一个audiotrack 对象:javaint sampleRateInHz = 44100;int channelConfig = AudioFormat.CHANNEL_OUT_MONO;int audioFormat = AudioFormat.ENCODING_PCM_16BIT;int bufferSizeInBytes =AudioTrack.getMinBufferSize(sampleRateInHz, channelConfig, audioFormat);AudioTrack audioTrack = newAudioTrack(AudioManager.STREAM_MUSIC, sampleRateInHz, channelConfig, audioFormat, bufferSizeInBytes,AudioTrack.MODE_STREAM);在上面的示例中,我们使用了AudioManager.STREAM_MUSIC 流类型,设置采样率为44100Hz,声道为单声道,采样位数为16 位。
第二单元过关检测(A卷)(时间:120分钟满分:150分)第一部分听力(共两节,满分30分)第一节(共5小题;每小题1.5分,满分7.5分)听下面5段对话。
每段对话后有一个小题,从题中所给的A、B、C三个选项中选出最佳选项。
听完每段对话后,你都有10秒钟的时间来回答有关小题和阅读下一小题。
每段对A.The man’s father.B.The woman’s sister.A.Their favourite food.B.The weekend plan.A.Having the first team practice.B.Collecting money.A.The baby keeps crying.B.Her mother moved in recently.(共15小题;每小题1.5分,满分22.5分)听下面5段对话或独白。
每段对话或独白后有几个小题,从题中所给的A、B、C三个选项中选出最佳选项。
听每段对话或独白前,你将有时间阅读各个小题,每小题5秒钟;听完后,各小题将给出5秒钟的作答时间。
每段对话或独白读两遍。
听第6段材料,回答第6、7题。
A.It is useful for his work.B.He has got a lot of money.A.The way of payment.B.The quality of the car.听第7段材料,回答第8、9题。
A.He ignored her.B.He argued with her.’s the relationship between the speakers?A.Husband and wife.B.Host and cleaner.听第8段材料,回答第10至12题。
A.In the police station.B.On the road.A.Talk to the people behind them.B.Pull over to the side of the road.A.They call the police.B.They go through their insurance.听第9段材料,回答第13至16题。
Unit 9 数字音频压缩Unit 9-1第一部分:MP3随着互联网时代的到来,希望通过电话线传输越来越多的信息。
音频信息是一种愈来愈多被下载的(多媒体)形式,无论是乐队的唱片选曲,无线电节目,还是视频伴音。
因为电话线的带宽有限,所以需要对信息(包括音频)进行压缩。
用于CD和数字电视中存储数字音频的传统方法是每秒抽取并记录一定次数的声音幅度值。
幅度值的精度是由用于存储幅度的比特位数决定的。
所以,音频信号消耗的带宽(或者内存)由以下三个因素决定:每秒钟的采样次数(频率),用于存储幅度的比特位数(比特深度)以及信号的长度(时间)。
当这三个参数已知时,很容易的计算出所用内存:内存=频率⨯比特深度⨯时间此外,如果信号是立体声的,内存就乘以二,因为立体声实际上用了两个信号。
这个等式能用来说明为什么在互联网上传输高品质音频信号时需要压缩。
CD音频采用44,100 Hz采样率的16比特立体声。
这就意味着1分钟的音频信号需要使用44,100⨯16⨯60⨯2 = 84,672,000比特,或略超过10兆字节。
一个标准的56 kbps的调制解调器需要84,672,000/57344 = 1477秒,或大约25分钟。
等待1分钟的音频要25分钟这么长的时间,因此必须有另一个选择,这个选择就是MPEG 音频第3层,或MP3。
编解码器人的耳朵仅能听到有限的频率范围,因此编解码可去除这个范围之外的所有声音。
因为这些声音是不可能被人听到的。
对声音应用一种心理声学模型。
在播放音调高的声音时,要提高使低频声能被听到的临界分贝数(分贝阈值)。
心理声学模型能去除所有通过这种方式“隐藏”的声音。
下一步就是联合立体声。
人脑无法估计低频声音的方向,所以这个阈值之下的声音都用单声道编码。
如果信号某些部分仍然高于所需的比特率,这些部分的音质就会下降。
最后,应用哈夫曼编码,该编码将所有比特码字根据其出现的频率换成独特的变长比特码。
例如:最常出现的比特模式编码成“01”,而次常出现的编码成“010”,下一个被编码成“011”,依此类推。
Setting Up an Audio Database for Music Information Retrieval Benchmarking Perfecto Herrera-Boyer Universitat Pompeu Fabra Pg. Circumval·lació 8 08003 Barcelona, Spain +34 935 422 806
perfecto.herrera@iua.upf.es
ABSTRACT In this white paper we summarize some general requirements and issues to be clarified in order to set up a usable database for audio-based MIR benchmarking. A broad-minded approach is required in order to go beyond pure retrieval issues and include other problems that underlie in the core or in the surroundings of Music Information Retrieval. We conclude with a proposal for calling up a specific task-force that is capable of setting up the needed database in the next twelve months.
1. INTRODUCTION In this paper we summarize some general requirements and issues to be clarified in order to set up a usable database for audio-based MIR benchmarking. Benchmark databases have become a powerful tool for other related areas to achieve the status of mature science. Besides the TREC database [9], the TIMIT [4] or SHATR [3] databases for speech recognition, and the UCI [1] database for machine learning can be cited here as examples (the interested reader may also check the NIST respository [5]). One of the main motivations for writing this position paper has been the realising that, among the Music IR test candidates listed by Byrd [2], no audio collection was present. We feel this document can be complementary to other White Papers such as those by Crawford and Brown [3], Melucci and Orio [6], and Pardo et al. [7].
2. CATEGORIES OF PROBLEMS We need to address the following problems: • Multi-level feature extraction (e.g. computing descriptors that can be used for solving some of the forthcoming problems)
• Segmentation (e.g. finding boundaries for homogeneous segments, that can be considered different than the others)
• Identification (e.g. telling if a given sound file corresponds to a given song “title” or if a given singer is singing in a piece of music)
• Classification (e.g. telling if a given sound file corresponds to a given genre, style or whichever enumerated-type based ontology have been defined for them)
• Structural description (e.g. telling if the music has parts that re-appear, if it has some mirroring or alternating organization of large blocks, etc.)
• Summarization (e.g. generating abridged versions of the original file that do not lose the essential music concepts that are present in the original)
• Retrieval by textual/numerical metadata (e.g. the user requests a sound from a given instrument taxonomy, or it requests files containing music with average tempos ranging from 130 to 160 b.p.m.’s)
• Retrieval by example (e.g. the user provides a file and requests to find a similar song, or the pervasive “query by humming/singing”)
As you can see, even though the “retrieval” aspect is the one to consider as the most instrumental for having this meeting and discussion, there are other related problems that go “beyond retrieval” and that would be benefit from a coordinated strategy from ours. Broadening our concerns to accommodate them, and not only considering the song similarity retrieval or the query by humming/singing cases as those defining the area of music information retrieval can be a key point in opening our initiative to other scientific communities such as those of musicologists, signal processing engineers, or computer music scholars, which have shared interests with ours.
3. TYPES OF FILES We can enumerate the following different types of files we need for a broad coverage of problems:
a) "sound samples” as isolated notes from acoustic or electronic instruments; this is the easiest type to be found; sounds from non-occidental music traditions should not be missed.
b) recordings from individual instruments (“phrases”, “loops” or “solos”): phrases are used for studying the automatic computation of rhythm description features (tick, beat, tempo), of expressivity resources (vibrato, rubato, accentuation, etc.),
c) complete polyphonic music pieces covering a wide range of genres, styles and orchestrations
All recordings should be in a “bare-minimum” format of monophonic 16 bit resolution and sampled at 44.1 KHz. Stereo, and high-resolution, higher-sampling rate recordings are also welcome, but some agreement on how to deal with conversions should be matter for discussions. Permission to make digital or hard copies of all or part of this