Moving-average-filter-3db

Moving-average-filter-3db

Stock-options-apple
Platform online-forex-trading
Sg-forex


Online-trading-academy-reviews-scottsdale Online-trading-demo-hdfc Option-trading-hdfc Making-money-option-trading Rahasia-of-biner-options-trading Moving-average-geeks-toy

Laser Measurement Thickness Displacement Distance Laser Triangulation Sensor Sensor triangulasi laser dapat dibagi menjadi dua kategori berdasarkan kinerja dan penggunaannya. Laser resolusi tinggi biasanya digunakan dalam aplikasi pemindahan posisi dan penempatan dimana akurasi tinggi, stabilitas dan drift suhu rendah diperlukan. Seringkali sensor laser ini digunakan dalam proses pemantauan dan sistem kontrol umpan balik loop tertutup. Sensor triangulasi triangulasi jarak jauh jauh lebih murah dan biasanya digunakan untuk mendeteksi keberadaan bagian, atau digunakan dalam menghitung aplikasi. Produk Menggunakan Sensor Triangulasi Laser Microtrak TM PRO 2D Seri sensor laser Microtrak TM PRO 2D menggunakan teknologi triangulasi laser CCD terbaru yang menawarkan pengukuran presisi kecepatan tinggi hingga 700Hz. Microtraktrade 3 Sensor resolusi tinggi, sensor perpindahan laser berkecepatan tinggi (sensor perpindahan linier non-kontak) menggunakan teknologi sensor CMOS terbaru yang bahkan menantang aplikasi pengukuran yang paling sulit sekalipun. Microtraktrade 4 Microtraktrade 4 adalah sensor laser terbaik untuk mengukur tinggi, ketebalan, perpindahan, getaran dan lainnya. Ini menyediakan data output dan power melalui satu kabel USB. Aplikasi Posisi Sensing Posisi umum adalah aplikasi yang paling umum digunakan untuk sensor laser. Respon cepat dan linier mereka membuat mereka ideal untuk aplikasi penempatan umpan balik statis dan aktif. Jarak operasi dan jangkauan pengukuran yang besar memberikan fleksibilitas untuk pemantauan proses dan pemantauan kualitas. Aplikasi tipikal meliputi: Perkerasan dan pengerjaan jalan beton Perutean jalur kereta api Lokasi robot Posisi pengelasan kepala Posisi dan pitch pada sirkuit terpadu Kontrol loop tertutup dari sistem robot dan penentuan posisi Pengukuran Dinamis Sensor non-kontak ideal untuk mengukur target yang bergerak karena mereka memiliki respons frekuensi tinggi. Dan jangan meredam gerakan sasaran dengan menambahkan massa. Sensor laser kami dirancang dengan frekuensi sampling 40 kHz dan respons frekuensi 20 kHz yang benar, menjadikannya ideal untuk aplikasi kecepatan tinggi seperti: Analisis run-out spindle Karakterisasi piezoelektrik Pengukuran getaran ultrasonik Pemantauan proses in-line Integritas segel vakum untuk industri pengalengan Profil permukaan berbagai bahan Pengukuran Ketebalan dan Dimensi Pengukuran ketebalan produksi on-line secara konvensional telah dibuat dengan menggunakan sistem pengukuran tipe kontak langsung. Sensor, seperti LVDTs, diposisikan di atas dan di bawah material yang diukur untuk melacak posisi permukaan. Output sensor digabungkan melalui perangkat lunak atau perangkat penjumlahan dan ketebalannya ditentukan. Sayangnya, metode tipe kontak menyebabkan masalah pengukuran. Tidak hanya bahan yang bisa diukur rusak tapi juga pemakaian sensor. Selain itu, sensor kontak lambat dan mungkin tidak melacak dengan tepat target yang mungkin bergerak atau bergetar, sehingga aplikasi ini ideal untuk sistem laser kita. Pengukuran ketebalan satu sisi dimungkinkan jika satu sisi bahan dapat ditahan konstan terhadap bidang referensi tetap, namun untuk hasil terbaik, dua sisi pengukuran lebih disukai. Ini karena pendekatan dua sisi menghilangkan kesalahan yang mungkin diperkenalkan dari materi yang bergerak atau bergetar. Pendekatan dua sensor kami menyamakan data sampling untuk kedua sensor, yang memastikan pembacaan ketebalan benar. Jenis sistem ini menyediakan keluaran analog (0-10V), (4-20 ma) dan digital (format biner RS-485). Entah bisa digunakan untuk memberikan hasil ketebalan, tapi analog adalah pilihan yang lebih disukai jika ketebalan frekuensi tinggi (gt100Hz) diperlukan. Aplikasi yang berhasil meliputi: Pemantauan proses ketebalan kayu Kontrol kualitas selama pembuatan blok beton Jarak pemisahan antara rol Ketebalan rotor rem Lembar dan ketebalan web Kelebihan Kunci Kemampuan untuk menyelesaikan pengukuran di bawah satu mikron, pada sebagian kecil dari biaya teknologi kinerja tinggi lainnya. Rentang pengukuran adalah yang memungkinkan berbagai persyaratan aplikasi Jarak operasi besar yang menyediakan daya tahan yang cukup untuk mengurangi kemungkinan kerusakan dari menghubungi target yang bergerak Bagaimana Sensor Triangulasi Laser Bekerja Sensor triangulasi laser mengandung sumber sinar laser solid-state dan detektor PSD atau CMOSCCD. . Sinar laser diproyeksikan pada target yang diukur dan sebagian balok dipantulkan melalui optik fokus ke detektor. Seiring target bergerak, sinar laser secara proporsional bergerak pada detektor. Sinyal dari detektor digunakan untuk menentukan jarak relatif terhadap target. Informasi ini kemudian biasanya tersedia melalui keluaran analog, antarmuka digital (biner) atau tampilan digital untuk pemrosesan. Prinsip triangulasi laser Perbedaan antara sensor CMOSCCD dan PSD Sensor tipe CMOS dan CCD mendeteksi distribusi puncak kuantitas cahaya pada array piksel sensor untuk mengidentifikasi posisi target, sedangkan sensor tipe PSD menghitung sentris balok berdasarkan titik pantulan keseluruhan pada sebuah array. Karena itu, sensor tipe PSD lebih rentan terhadap refleksi palsu dari perubahan kondisi permukaan, yang dapat mengurangi keakuratannya. Namun, saat mengukur ke ideal matte finish atau specular target resolusi mereka tidak ada bandingannya. Sistem CCD dan CMOS biasanya lebih akurat pada permukaan yang lebih luas karena hanya piksel berberat tertinggi dari sinar yang dipantulkan yang digunakan untuk menghitung posisi. Pita bermuatan rendah biasanya diberi energi oleh refleksi yang tidak diinginkan dari perubahan sifat optik permukaan yang diukur dan dapat dengan mudah diabaikan selama pemrosesan sinyal. Hal ini memungkinkan mereka untuk digunakan dalam berbagai aplikasi yang lebih luas. Gambar 2 menunjukkan perbedaan distribusi sinyal antara teknologi CMOS dan PSD, yang menyoroti masalah akurasi potensial yang terkait dengan sensor tipe PSD. Kesalahan potensial yang disebabkan oleh sensor laser tipe PSD Berlaku pada permukaan yang sangat reflektif atau cermin Sensor triangulasi laser juga dapat digunakan pada permukaan cermin atau reflektif yang sangat tinggi, yang biasa disebut permukaan specular. Dengan permukaan ini, sensor triangulasi khas, seperti yang ditunjukkan pada Prinsip Triangulasi Laser, tidak dapat digunakan karena sinar laser akan memantul langsung kembali ke dirinya sendiri. Untuk kasus ini diperlukan untuk mengarahkan balok ke target pada suatu sudut. Sinar akan memantulkan dari target pada sudut yang sama namun berlawanan dan fokus ke detektor. Kami memproduksi kepala laser yang dirancang khusus untuk permukaan specular atau laser kami dapat dipasang pada sudut dan dioperasikan dalam mode specular jika diperlukan. Prinsip operasi kepala laser specular Karakteristik Sensor Triangulasi Laser Sensor perpindahan Laser tanpa kontak tidak dihubungi berdasarkan disain. Artinya, mereka mampu secara tepat mengukur posisi atau perpindahan suatu benda tanpa menyentuhnya. Karena itu, objek yang diukur tidak akan terdistorsi atau rusak dan gerakan sasaran tidak akan dibasahi. Selain itu, sensor perpindahan laser dapat mengukur gerakan frekuensi tinggi karena tidak ada bagian sensor yang perlu tetap berhubungan dengan objek, membuatnya ideal untuk pengukuran getaran atau aplikasi lini produksi berkecepatan tinggi. RangeStandoff Distance Laser sistem triangulasi memiliki titik operasi yang ideal, yang kadang-kadang disebut sebagai jarak kebuntuan. Pada titik ini, laser berada pada titik fokus tertajam dan titik pantulan berada di tengah detektor. Seiring target bergerak, titik akan bergerak ke ujung detektor yang memungkinkan pengukuran pada rentang tertentu. Baik jarak dan jarak antara sensor ditentukan oleh desain optiknya. Kinerja optimal diperoleh pada jarak dekat karena titik paling kecil pada titik fokusnya dan sangat terkonsentrasi pada detektor. Algoritma pendeteksian benar untuk ketidakakuratan yang disebabkan saat beroperasi sedikit tidak fokus dan kebanyakan produsen menentukan kinerja selama rentang pengukuran yang lengkap. Untuk detektor panjang yang diberikan, sudut penerimaan yang lebih kecil menawarkan rentang pengukuran dan jarak operasi yang lebih besar. Sudut yang lebih besar memberikan kebalikannya, namun sensitivitas yang lebih tinggi dapat diperoleh karena adanya optical leveraging. Diagram disederhanakan ini memvisualisasikan perbedaan antara dua sensor sudut penerimaan yang berbeda Sensitivitas Dalam sistem pengukuran, sensitivitas biasanya ditentukan oleh berapa banyak perpindahan terjadi per unit pengukuran, biasanya dinyatakan dalam mikronmilli-volt. Semakin tinggi sensitivitas (digambarkan dengan angka yang lebih rendah) semakin baik dalam kebanyakan kasus karena resolusi yang lebih besar dapat diperoleh. Untuk mencapai sensitivitas tertinggi, ideal untuk memiliki sinar laser melintasi panjang detektor lengkap di atas rentang pengukuran aplikasi. Sensitivitas ditentukan oleh kemiringan respon output sensor. Output dari dua sensor dengan sensitivitas berbeda digambarkan dalam grafik. Harap dicatat bahwa kemiringan kurva masing-masing mewakili faktor sensitivitas masing-masing dengan Curve A dua kali lebih sensitif. Resolusi Resolusi sensor perpindahan laser didefinisikan sebagai jumlah terkecil perubahan jarak yang dapat diukur dengan andal. Bila dirancang dengan benar, sensor triangulasi laser menawarkan resolusi dan stabilitas yang sangat tinggi, seringkali mendekati sistem interferometer laser mahal dan kompleks. Karena kemampuan mereka untuk mendeteksi gerakan kecil seperti itu, mereka telah berhasil digunakan dalam banyak aplikasi presisi presisi tinggi. Faktor utama dalam menentukan resolusi adalah sistem noise listrik. Jika jarak antara sensor dan target konstan, output akan tetap berfluktuasi sedikit karena white noise pada sistem. Diasumsikan bahwa, tanpa pemrosesan sinyal eksternal, seseorang tidak dapat mendeteksi pergeseran output kurang dari kebisingan acak instrumen. Karena ini, sebagian besar nilai resolusi disajikan berdasarkan nilai kebisingan puncak-ke-puncak dan dapat ditunjukkan dengan rumus spesifik: Sensitivitas Resolusi x Kebisingan Berdasarkan rumusan, jelas bahwa untuk sensitivitas tetap, resolusi hanya bergantung Atas kebisingan sistem. Semakin rendah noise semakin baik resolusinya. Jumlah noise tergantung pada bandwidth sistem. Hal ini karena kebisingan umumnya didistribusikan secara acak melalui berbagai frekuensi dan membatasi bandwidth dengan penyaringan akan menghilangkan beberapa fluktuasi frekuensi yang tidak diinginkan. Sensor laser kami juga memberikan nilai perpindahan dalam format digital. Resolusi keluaran digital dihitung dengan membagi rentang perpindahan dengan bit rate prosesor. Sebagai contoh, sebuah sensor dengan kisaran mikron 2000 akan memiliki resolusi 20002E16, atau 0,03 mikron untuk sistem 16 bit. Jika menggunakan konverter 12 bit, resolusi akan lebih buruk pada 20002E12, atau 0,5 mikron. Angka di bawah ini menunjukkan perbedaan output dari dua sistem yang sama dengan low pass filters yang berbeda. Semua sistem triangulasi laser kami memiliki perangkat lunak low pass filter yang mudah disesuaikan untuk memudahkan penyesuaian di lapangan. Amplifier output noise dengan 20kHz low pass filter Amplifier output noise dengan 100Hz low pass filter Bandwidth, atau frekuensi cutoff, dari suatu sistem biasanya didefinisikan sebagai titik dimana output dibasahi oleh -3dB. Ini kira-kira sama dengan penurunan voltase keluaran 30 dari nilai sebenarnya. Dengan kata lain, jika target bergetar dengan amplitudo 1mm pada 5 kHz, dan bandwidth sensor laser diatur pada 5 kHz, output sebenarnya adalah 1mm X 70 0.70mm. Jadi, penting untuk mengatur respon frekuensi sistem lebih tinggi dari pada gerakan target yang diharapkan. Semua sensor laser kami memiliki pengaturan filter yang dapat disesuaikan. Filter yang sesuai harus dipilih agar aplikasi mencegah atenuasi output. Teknisi aplikasi kami dapat membantu dalam memilih pengaturan filter yang sesuai. Resolusi Spasial Saat melakukan pengukuran, sensor laser memberikan jarak kira-kira sama dengan lokasi permukaan rata-rata di dalam titik laser. Mereka tidak mampu mendeteksi secara akurat posisi fitur yang lebih kecil dari ukuran titik, namun mereka dapat berulang kali mengukurnya pada permukaan yang kasar. Karena itu, titik laser harus selalu kira-kira 25 lebih kecil dari fitur terkecil yang ingin Anda ukur. Bintik yang lebih kecil bisa membedakan fitur yang lebih kecil pada objek. Dalam dunia ideal, output dari sensor manapun akan sangat linier dan tidak menyimpang dari garis lurus pada titik apapun. Namun, pada kenyataannya akan ada sedikit penyimpangan dari garis ini, yang mendefinisikan sistem linearitas. Biasanya, linearitas ditentukan sebagai persentase dari Rentang Pengukuran Skala Penuh (Full Scale Measurement Range / FSR). Selama kalibrasi, output dari kepala laser dibandingkan dengan keluaran dari standar yang sangat presisi dan perbedaan dicatat. Perbedaan ini secara otomatis dikoreksi karena penggunaan tabel look up. Sensor laser Microtrak II kami menawarkan linieritas tertinggi yang ada saat ini. Sebagian besar sistem melebihi -0,05 FSR dengan beberapa pencapaian -0,01 atau lebih baik. Akurasi adalah fungsi linearitas, resolusi, stabilitas suhu dan drift, dengan linearitas menjadi penyumbang utama. Respon linier sensor kita sangat berulang. Laporan kalibrasi memberikan data yang dapat digunakan untuk mengoreksi non-linearitas suatu sistem dengan komputer murah dan perangkat lunak koreksi, yang menghasilkan akurasi yang lebih baik jika diperlukan. Menerapkan Sensor Triangulasi Laser Material dan Finish Saat menerapkan sensor laser, pertama-tama perlu untuk menentukan reflektivitas permukaan. Hasil akhir matte yang konsisten diinginkan untuk performa terbaik saat menggunakan kepala diffuse. Jika finishing yang sangat dipoles atau cermin akan digunakan, kami sangat merekomendasikan kepala laser specular. Target Shape Untuk performa ideal, targetnya harus diposisikan pada 90 derajat ke kepala laser untuk mencegah kesalahan kemiringan. Pengaruh dari kemiringan akan tergantung pada sifat reflektif permukaan. Target ideal yang menyebar memungkinkan operasi yang tepat pada permukaan miring 30 derajat atau lebih dari normal. Namun, target cermin akan menghasilkan kesalahan jika kemiringan berubah sesedikit 1 derajat. Perhatian harus dilakukan selama desain dan operasi fixture untuk meminimalkan kemiringan target. Sensor laser juga bisa digunakan untuk mengukur target melengkung. Untuk hasil terbaik, balok harus diposisikan menghadap langsung ke arah pusat kelengkungan. Ini hampir menghilangkan kemiringan yang dilihat oleh laser. Selain itu, orientasi kepala harus sedemikian rupa sehingga permukaan melengkung tidak condong pada sudut triangulasi laser. Gambar di bawah menunjukkan orientasi yang tepat pada sistem untuk mengurangi efek kemiringan. Perhatikan bagaimana sinar laser bisa dibelokkan dengan bentuk target. Sadar akan fitur target Anda sebelum mengukur untuk memastikan sinar kembali laser tidak terhalang. Gambar di bawah menunjukkan cara yang benar dan salah untuk mengorientasikan sensor laser. Kondisi Lingkungan Karena sistem triangulasi laser adalah sensor tipe optik, penting untuk menjaga agar jalur optik tetap bersih dan bebas dari penghalang atau bahan asing. Kotoran, debu dan asap dapat mempengaruhi hasil pengukuran atau bahkan membuat sensor sama sekali tidak berguna. Perhatian harus dilakukan untuk menghilangkan kontaminasi dan pembersihan sistem pembersih udara yang bersih harus digunakan bila diperlukan. Jika jenis sistem ini tidak memungkinkan, penting untuk secara teratur membersihkan lensa luar untuk menghindari komplikasi. Masalah lingkungan yang paling umum yang dapat mempengaruhi keakuratan sensor laser adalah suhu. Tidak hanya pameran elektronik yang menyapu suhu, namun juga perluasan dan kontraksi komponen mekanis dan fixturing dapat mengubah celah sensor secara fisik. Semua sensor Microtrak II kami memiliki stabilitas suhu kurang dari -0,05 dari rentang pengukuran skala penuh selama perubahan suhu 0 sampai 40oC. Penting agar fixture memegang sensor triangulasi laser stabil. Menimbang bahwa perubahan suhu dapat menyebabkan ekspansi dan kontraksi, sehingga terjadi perubahan jarak pada target, perlengkapan harus dibuat dari bahan yang sesuai untuk meminimalkan efek ini. Dukungan fixture juga harus sesingkat mungkin dan cantilevers lama harus dihindari untuk meminimalkan tidak hanya masalah suhu tapi juga mengurangi getaran. Sensor laser kami memiliki lubang yang bisa digunakan untuk memasang dan mengamankan kepala laser. Jadwal harus dilakukan agar sesuai dengan lokasi lubang ini dan menjaga kepala laser tegak lurus terhadap sasaran yang diminati. Sinkronisasi Saat membuat pengukuran ketebalan diferensial dengan 2 kepala laser, penting untuk mengambil dan memproses pengukuran dari kedua kepala pada waktu yang bersamaan. Prosedur ini menghilangkan hasil yang tidak diinginkan akibat getaran. Jika targetnya bergerak, dan pengukuran dilakukan pada waktu yang sedikit berbeda, hasil yang diproses mungkin melaporkan target yang sedikit lebih tipis atau lebih tebal. Garis sensor laser Microtrak II kami memiliki ketentuan untuk menyinkronkan kepala yang menghilangkan masalah ini. Pada bab ini, kami memeriksa peralatan dan teknik pemahasan kebisingan serta pengembangan survei peta, dan beralih ke interpretasi hasil yang sederhana. Pengukuran Kebisingan dan Survei Mengapa kita mengukur tingkat suara Pengukuran suara memungkinkan analisis rinci dan tepat untuk semua suara. Dalam kasus suara menyenangkan ini mungkin berarti bahwa kita dapat meningkatkan jumlah kesenangan yang terkait dengannya. Contohnya adalah memperbaiki kinerja komponen sistem Hi-Fi, atau memperbaiki akustik auditorium. Ketika pengukuran suara diterapkan pada suara yang mengganggu atau tidak diinginkan, hal itu digunakan untuk mencoba dan mengurangi kejengkelannya. Karena karakterisasi kebisingan merupakan area yang sangat subjek, tidak semua pihak akan sepakat mengenai tingkat gangguan sumber kebisingan. Memang perbedaan psikologis dan fisiologis antara manusia hampir memastikan bahwa pengukuran ilmiah tidak akan dapat sepenuhnya menilai apa yang setidaknya sebagian merupakan penilaian subjektif. Namun, pengukuran tidak memberi kita sarana untuk membandingkan suara dalam kondisi yang berbeda. Mereka juga memberikan informasi yang jelas tentang kapan suara cenderung merusak pendengaran, dan oleh karena itu disarankan bila tindakan perbaikan seperti peralatan pelindung diperlukan. Selain itu, kehilangan pendengaran dapat dinilai, untuk memperkirakan jumlah kerusakan yang dilakukan pada telinga orang. Hal ini dilakukan melalui pengujian audiometrik, yang mengukur kinerja pendengaran individu di sejumlah rentang frekuensi. Pengujian audiometrik merupakan komponen penting dari program konservasi pendengaran yang berhasil. Pengukuran kebisingan juga penting untuk program pengurangan kebisingan. Area kebisingan yang tinggi seperti pabrik, bandara, jalan yang sibuk, tambang dan pusat hiburan harus dievaluasi secara teratur untuk menilai dampaknya terhadap lingkungan. Apa Perbedaan antara Suara Suara dan Tekanan Suara Istilah-istilah ini biasa digunakan dalam penilaian kebisingan, namun sering kali salah digunakan. Daya suara mengacu pada jumlah energi suara yang ditransmisikan oleh sumber suara. Tekanan suara adalah perbedaan antara tekanan aktual yang dihasilkan oleh gelombang suara dan tekanan rata-rata atau barometrik pada titik tertentu di ruang angkasa. Unit lain, sound level juga biasa digunakan. Ini mengacu pada tingkat tekanan suara tertimbang yang diperoleh dengan menggunakan meteran tingkat suara dan jaringan pembobotan frekuensi, seperti A, B, atau C. Ini akan dibahas lebih rinci nanti di bab ini. Sebuah analogi yang baik untuk istilah-istilah ini dapat diperoleh dengan mempertimbangkan sebuah langkah kaki memasuki kolam air. Kekuatan suara sama dengan jumlah energi yang digunakan kaki untuk mengatur riak di kolam, sedangkan tekanan suara adalah seberapa banyak perahu yang mengambang di kolam bergerak ke atas dan ke bawah oleh riak-riak. Pengukuran tingkat tekanan suara Pengukuran suara dan kebisingan dilakukan dengan menggunakan meteran tingkat suara dimana ada beberapa jenis yang tersedia. Mereka dirancang untuk merespons suara dengan cara yang hampir sama seperti telinga manusia, dan memberikan evaluasi tingkat tekanan suara yang obyektif dan dapat direproduksi. Sebagian besar didasarkan pada prinsip sederhana menggunakan mikrofon sebagai alat deteksi yang memberi dorongan listrik ke amplifier yang kemudian melewati sinyal ke meter digital (biasanya dikalibrasi di dB). Pengukuran biasanya diambil selama periode waktu standar (0,1-1,0 detik). Analisis Frekuensi dan Bobot Suara (A, B, C, D, Lin) Meteran tingkat suara yang lebih mahal dapat diatur untuk menganalisis frekuensi gelombang suara, atau intensitas masing-masing frekuensi yang berbeda yang membentuk kebisingan. Selain itu, sinyal suara dapat dilewatkan melalui jaringan pembobotan, yang merupakan rangkaian elektronik yang sensitivitas terhadap frekuensi suara berbeda berbeda dengan telinga manusia. Ini mensimulasikan kontur kenyaringan yang sama. Pada bab sebelumnya, terlihat bahwa kenyaringan suara yang jelas terkait dengan frekuensi suara dan juga tingkat tekanan suara. Ini karena respon suara manusia terhadap suara bervariasi sesuai dengan frekuensi gelombang suara. Peralatan pengukuran suara sekarang dirancang untuk membuat tunjangan untuk perilaku telinga dengan menggunakan jaringan pembobot elektronik, yang menyaring frekuensi suara tertentu, dan mendukung yang lain. Ada beberapa karakteristik standar internasional yang berbeda yang digunakan untuk suara 8220weight8221. Ini disebut pembobotan A, B, C, D, dan Lin. Rangkaian pembobotan A dirancang untuk mendekati respons telinga manusia rata-rata pada tingkat tekanan suara rendah. Demikian pula sirkuit pembobotan B dan C dimaksudkan untuk mendekati respons telinga manusia masing-masing pada 55 8211 85dB dan 85 dB. Karakteristik dari jaringan ini ditunjukkan pada Gambar 2.1. Bobot lainnya, bobot D terkadang digunakan untuk kebisingan pesawat. Sebagian besar sound level meter juga memiliki jaringan linier. Ini sama sekali tidak memberi bobot pada sinyal, namun membiarkan sinyal melewati tidak dimodifikasi. Tingkat suara yang diperoleh dengan cara ini disebut Lin. Paling umum akhir-akhir ini jaringan bobot A digunakan 8211 dan kecenderungannya mendekati penggunaan hampir secara eksklusif bila jaringan pembobotan diinginkan. Ini karena jaringan pembobotan B dan C tidak berkorelasi baik dengan tes mendengar subjektif. Ini mungkin karena kontur kenyaringan yang sama yang digunakan untuk mengatur skala bobot menggunakan nada murni 8211 dan sebagian besar suara alami terdiri dari sinyal kompleks yang terbuat dari berbagai nada. Gambar 2.1 8211 Kurva pembobotan A, B, dan C untuk meter tingkat suara (dari Bies amp Hanson) Tabel 2.1 menunjukkan koreksi pada dB yang dibuat pada frekuensi tertentu untuk mendapatkan bobot A. Tabel 2.1 8211 Koreksi (dB) dibuat pada frekuensi untuk mendapatkan dB (A) 1 Dimungkinkan untuk menggunakan nilai dari tabel 2.1 untuk mengubah tingkat suara yang tidak tertimbang (Lin) di dB menjadi tingkat suara tertimbang A seperti ditunjukkan pada contoh Diberikan di bawah pada Tabel 2.2. Tabel 2.2 8211 Contoh penerapan pembobotan A ke tingkat suara linier1. Tingkat suara tertimbang A ditemukan dengan menambahkan koreksi tertimbang A ke nilai suara linier untuk setiap frekuensi. Ini menghasilkan nilai yang ditunjukkan pada baris 3. Nilai tingkat suara pada baris 3 kemudian ditambahkan untuk memberikan tingkat suara tertimbang A dalam unit yang disebut dB (A). Bila ditambah bersama hasil akhir adalah 84.9dB (A), bukan 650 yang ganjil. Hal ini dijelaskan kemudian pada bagian tentang penggabungan pembacaan desibel. The (A) di sini hanya berarti bahwa tingkat suara telah diperoleh baik dengan menggunakan pembobotan jaringan A pada meteran suara atau telah diubah menjadi pembobotan A seperti yang ditunjukkan di atas. Menggabungkan Tingkat Suara dalam Decibel Seringkali perlu untuk menggabungkan pengukuran dari beberapa sumber suara yang dinyatakan dalam desibel untuk mendapatkan tingkat kebisingan total. Misalnya kita mungkin diminta untuk memprediksi tingkat kebisingan di tempat kerja jika mesin tingkat emisi suara yang diketahui ditambahkan. Hal ini bisa dilakukan dengan menambahkan level suara dari mesin ke tingkat background yang sudah ada di tempat kerja. Tidak bisa dilakukan hanya dengan menambahkan nilai di dB. Ini karena skala dB adalah skala logaritmik yang dibikin. Untuk menggabungkan tingkat suara, penting untuk mengkonversikannya kembali ke tingkat energi, menambahkannya, lalu mengkonversikannya kembali ke dB. Bila ini dilakukan, Anda akan menemukan bahwa menggabungkan sumber suara 80dB dengan suara 80dB lainnya akan menghasilkan tingkat tekanan suara total 83dB bukan 160dB yang diharapkan. Dengan kata lain setiap 3dB meningkatkan tingkat tekanan suara dua kali lipat. Matematika menggabungkan nilai dB sedikit rumit. Tanpa mencoba menjelaskan bagaimana hal tersebut dilakukan persamaan untuk perhitungannya adalah sebagai berikut. Tingkat tekanan suara 10log 10 10 (nilai dB 1 10) 10 (nilai dB 2 10) 10 (nilai dB ke-3 10). Misalnya jika Anda ingin menambahkan suara dari tiga sumber suara dengan tingkat tekanan suara 89dB, 90dB dan 91dB, Anda akan melanjutkannya sebagai berikut. Tingkat tekanan suara 10log 10 10 8910 10 9010 10 9110 10log 10 7.94 x 10 8 10 x 10 8 12,59 x 10 8 10log 10 30,53 x 10 8 Catatan: Nilai ini diperoleh dengan membagi indeks (misalnya 8910 untuk mendapatkan 8,9), Kemudian mengambil anti-log (push inverse kemudian log on calculator) hasilnya. Dalam hal ini adalah 7,94 x 10 8. Jika Anda ingin mengurangi tingkat suara menggunakan persamaan yang sama kecuali bahwa nilai dalam tanda kurung siku dikurangi. Contohnya mungkin jika Anda memiliki mesin berisik di pabrik dan ingin mengetahui suara dari mesin saja. Dengan mesin pada tingkat kebisingan di pabrik adalah 95dB, namun bila dimatikan maka akan berkurang menjadi 89dB. Suara dari mesin ditentukan sebagai berikut: Tingkat tekanan suara 10log 10 10 9510 8211 10 8910 10log 10 31.62 x 10 8 - 7.94 x 10 8 10log 10 23.68 x 10 8 Kalkulator Kebisingan 1 - Menambahkan amp Mengurangi Tingkat Tekanan Suara Masukkan nilai Anda ingin menghitung ke dalam sel SPL dan tab melalui. Jawabannya akan muncul secara otomatis Jika Anda menemukan semua matematika ini terlalu keras, ada grafik yang dapat digunakan untuk menyimpulkan tingkat suara gabungan dari dua sumber. Hal ini ditunjukkan pada gambar 2.2. Gambar 2.2 8211 Bagan untuk menggabungkan tingkat suara Untuk menggunakan ini untuk memperkirakan efek menggabungkan dua sumber suara 8211 satu dari 88dB dan yang lainnya 91dB, lanjutkan sebagai berikut. Temukan perbedaan antara dua tingkat suara Temukan nilai ini pada sumbu X dari Chart Proceed langsung naik dari sumbu X sampai memotong kurva. Pindahkan lurus ke sumbu Y dan catat nilainya Tambahkan nilai yang diperoleh dari sumbu Y ke sumber yang ribut Untuk mendapatkan tingkat kebisingan beberapa sumber gabungan, ulangi lima langkah pertama untuk sumber kebisingan tingkat terendah, lalu hitung kombinasi mereka dengan Sumber noisiest berikutnya Misalnya jika Anda ingin menambahkan suara dari tiga sumber suara dengan tingkat tekanan suara 87dB, 90dB dan 94dB, Anda akan melanjutkan sebagai berikut Dua sumber gangguan tingkat terendah adalah 87 dan 90, oleh karena itu perbedaannya adalah 3dB Nilai yang dikoreksi dari Grafik adalah 1.7dB (lihat garis putus-putus pada Gambar 2.2) Total kebisingan dari kedua sumber ini adalah 90 1,7 91.7dB Perbedaan tingkat kebisingan antara mesin paling keras dan suara gabungan dua lainnya adalah 94 8211 91.7 2.3dB Nilai koreksi dari grafik Adalah 2.0dB Total suara dari kedua sumber ini adalah karena itu, dengan demikian dapat memberikan bobot suara yang benar ke tingkat persepsi telinga manusia, atau untuk mendapatkan informasi lebih terperinci tentang suara kompleks itu. Diperlukan untuk membagi rentang frekuensi bunyi yang dapat didengar (20 8211 20,000Hz) ke dalam band. Hal ini dilakukan dengan menggunakan filter elektronik yang menolak semua suara dengan frekuensi di luar band yang dipilih. Band ini biasanya memiliki lebar 13 dari oktaf atau 1 oktaf. Bagi mereka yang tidak terbiasa dengan musik, oktaf adalah frekuensi dua kali lipat (yaitu dari 260 menjadi 520Hz adalah satu oktaf). Pada piano ini berarti menggerakkan delapan kunci putih (maka istilah oktaf). Pada grafik frekuensi suara itu berarti pita frekuensi dimana frekuensi yang lebih tinggi dua kali frekuensi lebih rendah. Dalam pengukuran suara, nilai yang dinyatakan untuk pita frekuensi biasanya adalah frekuensi pusat. Ini berarti bahwa berbagai suara diperbolehkan melalui filter suara dengan frekuensi yang dinyatakan sebagai pusatnya. Contoh dari hal ini adalah frekuensi pusat 1000Hz. Di band ini filter memungkinkan semua suara frekuensi antara 707 8211 1414Hz melaluinya, namun menolak yang lainnya. Proses membagi suara kompleks menjadi band disebut analisis frekuensi, dan hasil analisis frekuensi disajikan pada bagan yang disebut spectrogram atau histogram frekuensi. Gambar 2.3 menunjukkan spektogram bentuk gelombang kompleks. Gambar 2.3 8211 Spektogram dari bentuk gelombang kompleks (dari Bruumlel dan Kjaeligr 8211 Measuring Sound). Setelah sinyal dibagi menjadi pita frekuensi, maka bobotnya, umumnya diperkuat di dalam meteran tingkat suara dan nilai Root Mean Square atau (RMS) yang ditentukan dengan menggunakan detektor RMS. Ini menentukan jumlah energi dalam suara yang diukur. Mengukur Paparan Kebisingan 8211 Tingkat Suara Kontinu yang Setara (Leq) Seiring gelombang suara mentransfer energi, jumlah potensi kerusakan pendengaran yang terkait dengan lingkungan yang bising terhubung langsung ke tingkat suara dan jumlah waktu seseorang terpapar. Itu adalah pemaparan satu jam terhadap suara keras akan lebih berbahaya bagi pendengaran kita daripada satu menit yang terpapar tingkat tekanan suara yang sama. Pemaparan Kebisingan x Kerusakan Waktu Waktu Untuk menilai potensi kerusakan pendengaran pada lingkungan yang bising, tingkat tekanan suara dan waktu pemaparan harus diukur. Hal ini memungkinkan kita untuk menentukan jumlah energi yang diterima oleh telinga. Ketika tingkat suara cukup konstan, ini adalah proses yang sederhana, namun jika berfluktuasi maka tingkat suara harus diambil sampel berulang kali selama periode sampling yang ditetapkan agar mendapatkan hasil yang valid. Hal ini kemudian memungkinkan untuk memperoleh satu nilai yang disebut tingkat suara kontinyu setara (Leq) yang menghubungkan jumlah energi yang diterima dari waktu ke waktu dari suara yang berfluktuasi ke tingkat pemaparan tingkat kontinu yang setara. Oleh karena itu, ini berkaitan dengan potensi relatif kerusakan pendengaran terhadap dosis kebisingan kontinu yang setara. Contoh yang baik dari waktu yang bervariasi adalah perkiraan kebisingan lalu lintas di daerah perkotaan. Bila pengukuran tingkat suara juga A-weighted dosisnya disebut LAeq. Selain menentukan potensi kerusakan pendengaran suara, jenis pengukuran ini juga umum digunakan dalam penilaian gangguan kebisingan masyarakat. In order to be able to calculate the LAeq of a noise source it is essential to use an integrating sound level meter (see section on sound level meters) which accumulate and average the noise dose over a set time period 8211 usually 60 seconds. LAeq values can also be used to estimate the noise dose to a person who moves around in a work environment where noise levels vary greatly. For example if a worker spends one hour in a very noisy workshop (at 100dB(A)), then the other seven hours behind a desk in a quiet office (at 70dB(A)), their equivalent noise exposure level would be 91dB(A). Figure 2.4 8211 Estimation of LAeq values (from Bruumlel and Kjaeligr 8211 Measuring Sound2) Equivalent sound can also be useful in evaluating community noise over a 24hr period. Figure 2.5 shows how this may be used. The dotted line indicates the Leq for the same period, giving an indication of the constant sound level that would have delivered equivalent sound energy to the ears of residents. Figure 2.5 8211 Use of equivalent sound level for evaluating community noise These are the most common instruments used to measure sound pressure levels. The main components of a typical sound level meter are shown in Figure 2.6. Although there are many different brands and types of sound meters available, all have the basic layout shown. The microphone is a transducer that converts a physical measurement of sound waves into an electrical signal. The voltage that leaves the microphone is proportional to the sound pressure level. The most suitable type of microphone for sound level meters is the condenser microphone, which is very stable and reliable, but only produces a small voltage. This means that a preamplifier is needed to boost the signal before it can be processed. There are several different types of processing that may be performed on the signal. These will vary according to the type of assessment being conducted. The simplest type of processing involves passing the signal straight through to the detector unmodified. This produces the Lin or linear sound rating. It is sometimes referred to as an all pass network. More often though the signal is passed through a weighting network. The function of these has already been discussed, in previous sections. Basically they are just selective electronic filters which provide A, B, or C weightings to the sound frequencies. After the sound levels are weighted they are passed on to the amplifier, then the RMS detector, which accurately assess the amount of energy transferred by the sound wave. Figure 2.6 8211 The components of a typical sound level meter (above) Figure 2.7 8211 A typical hand held sound meter (from Bruumlel and Kjaeligr Instruments) Detector modes for sound meters Sound waves are rarely steady in level. In order to cope with the often large fluctuations of sound levels found during analysis, most sound level meters are provided with two or more response modes. Some of these include slow response mode, fast response mode, impulse mode and peak mode. This means that the meter responds much like the human ear 8211 in approximately 100 8211 125msec. It allows accurate analysis of sound that does not fluctuate very rapidly. This mode uses a time frame of approximately one-second to average the sound level. This does not mimic the response of the human ear, but rather is used to evaluate sound levels with large fluctuations in intensity during measurement. This is a very rapid response mode (typically 35msec). It is used to analyse very short sharp sounds such as dropping objects and impacts which could not be accurately followed by typical sound meter settings. This is referred to as transient noise. In this mode the meter responds like the human ear to transient noise peaks. Although the perceived loudness of very short duration sound is lower than that of steady continuous sound, this does not mean that the potential damage to hearing is reduced. For this reason most sound level meters include a circuit for measuring the peak value of the sound. This is the highest sound pressure level reached during measurement 8211 regardless of the duration for which it occurred. This is an important parameter in most occupational noise exposure standards. In Australia peak values are referred to as short-term exposure levels or STEL. Most meters also include a special circuit called a Hold circuit which stores the Peak value or the maximum RMS value. Integrating Sound Level Meters Noise levels encountered in practice are rarely constant in level. Often the fluctuations in sound pressure levels are quite large. In order to measure these levels most high quality sound meters are equipped with an integrating facility, which allows mean sound levels over a set time period to be established. More detail on this is given in the section on equivalent sound levels Calibration of Sound Level Meters As with all scientific instrumentation sound level meters must be calibrated to enable accurate and reliable sound level measurement. A complete procedure involves both electrical calibration and acoustic calibration. This involves the use of an internal oscillator of known frequency which checks the amplifier, the weighting networks and the output meter. If any of these are incorrect they may be adjusted by controls on the meter. This form of calibration does not however check the performance of the microphone, which must be checked by regular acoustic calibration. This involves placing a small acoustic calibrator (sometimes called a pistonphone) on the microphone, and comparing the result with the known reference value of the calibrator. These devices provide precisely defined sound pressure levels to which the sound pressure level meter should be adjusted (for example one provides a 94dB output at 1,000Hz). Generally accuracy of up to 0.2dB can be obtained by using these calibrators and a good quality meter. This calibration is normally limited to a few discrete frequencies, so it is by no means totally foolproof. It is good technique to calibrate a meter both before and after a sound measurement session to ensure that valid results have been obtained. Any large errors indicate damage to either the sound meter or the calibrator 8211 in which case both should be serviced. Common Errors in Performing Sound or Noise Measurements As with most scientific instruments, unless sound pressure level meters are used correctly, then the data produced with them is meaningless. Some of the more important points with regard to their use are listed below. In general the most important sources of error in obtaining sound levels include mishandling of microphones and meters, wind, temperature, dust, humidity, changes in ambient pressure, vibrations, magnetic fields, and background noise. Mishandling of microphones and equipment If microphones are placed on surfaces that are vibrating, then the microphone will produce signals that register on the meter as sound pressure. Exposure to extremely high noise levels and dropping microphones will also cause great changes in calibration of the instrument. Even a light breeze blowing across the microphone will produce spurious noise. This sounds somewhat like someone blowing in your ear and is principally made up of frequencies below 200Hz. Fitting a porous foam windscreen over the microphone reduces these effects. Additionally this protects the microphone from dust and humidity. Most commercially available windscreens do not work effectively in winds above 20kmh and they do attenuate (reduce) the levels of high frequency sound. If sound levels must be obtained in windy conditions then special acoustic enclosures may be used. Relative humidity levels of up to 90 have little or no effect on noise measurements. Moisture does however affect the long-term performance of the microphone. For this reason they should always be protected from rain and stored in a dry (or desiccated) environment. Most sound level meters are designed to work in temperature ranges from 821110 - 50degC. So temperature is not normally a problem. If the instrument is moved from an air-conditioned environment into a hot environment however, condensation forming on the microphone may cause problems. Very low temperatures may cause battery failure. Variations in atmospheric pressure of up to 10 cause little error (0.2dB) on microphone sensitivity, but at high altitudes the sensitivity of the instrument to high frequencies drops off. To get around this special adjustments must be made when calibrating with the pistonphone. Reflected and absorbed sounds Objects near sound sources may greatly affect the sound output. For the purpose of measurement to assess things such as operator levels, these must be left in place to ensure an accurate reflection of the true environment is obtained. If the true sound power output of a particular machine is required however, it may be necessary to remove such items. One factor that is often overlooked in sound measurement of noise sources is the level of background noise compared to the level of sound being measured. The background noise must be such that it does not overwhelm the sound being generated by the noise source. What this means in practice is that the level of the sound being measured must be at least 3dB higher than background for any sort of measurement to be made. This means that it must be at least twice as loud. Even when there is 3dB or more difference between background and the sound source being measured, it is still generally necessary to apply a correction factor to obtain a valid result. This is done by following the procedure below Measure the total noise level (LSN) Measure the background level only (LN) by turning off the noise source 8211 if possible. Find the difference between the two levels (LSN - LN). If the difference is less than 3dB then the background level is too high for reliable measurement. If it is between 3 8211 10 dB then a correction factor must be applied. If it is greater than 10dB then correction factors are unnecessary. An example of the use of correction factors is given in Figure 2.8. Use the correction chart to estimate the correction factor. The value for (LSN - LN) is plotted in the x-axis, and its intersecting value from the Y 8211axis obtained. The value obtained from the y-axis is subtracted from the total noise level (LSN). Figure 2.8 8211 Applying a correction for background noise. 1 Bies, D.A. and Hanson, C.H. Engineering Noise Control 8211 Theory and Practice, Chapman amp Hall, London, 1996. 2 Bruumlel and Kjaeligr, Measuring Sound, Naeligrum, 1984. 3 NSW DECC, Stationary Noise Source Policy, decc.nsw.gov.au, 1999 or later. 4 Standards Australia, AS 1055 8211 Acoustics 8211 Description and Measurement of Environmental Noise, 1997. 5 Standards Australia, AS 1217 8211 Acoustics 8211 Determination of Sound Power Levels of Noise Sources, 1985. 6 Standards Australia, AS IEC 61672.1 8211 Electroacoustics Sound Level Meters, 2004Glossary of Terms The following is an extensive list of terms that you may or may not be familiar with that you may encounter at our site or throughout your audio experience. If nothing else. use a few every now and then so everyone thinks youre really smart This Glossary is borrowed from the Diamond Cut MillenniumLIVE manual. any reference to those programs is purely intentional. Acoustical impedance is the total opposition provided by acoustical resistance and reactance to the flow of an alternating pressure applied to a system. More specifically, it is the complex quotient of the alternating pressure applied to a system by the resulting volume current. The unit is the acoustical ohm. Acoustical reactance is the imaginary part of the acoustical impedance. Energy is not dissipated by acoustical reactance it is only stored there. The unit is the acoustical ohm. Acoustically mastered record recordings utilized only the energy of the sound waves created by the sound source to modulate the master cutting lathe stylus. This recording technique had none of the benefits which signal amplification can provide to the recording process. This is the method that was utilized from the time of the invention of the phonograph by Thomas Edison in 1876 up until around 1925, when vacuum tube amplifiers and microphones began to be employed in the mastering process. By 1929, all of the major record companies had switched over to the quotelectrical processquot of record mastering. Acoustical Resistance is the real term of the acoustical impedance relationship. This is the term responsible for the dissipation of energy. The unit is the acoustical ohm. A device used to convert analog signals into digital (discrete time) signals, so that they can be signal processed by a computer algorithm. The sound card in your computer contains an A-D converter and also a D-A (Digital to Analog) converter. To be compatible with DC MillenniumLIVE . it must have at least 16-bit resolution. However, the software does support 8 through 24 bit resolution sound cards. (216 65,536). In other words, your sound card must be able to divide the amplitude of audio signals into numerically sampled representations, the smallest division being one part in 65,536. 16 bit audio has the same resolution as red book CD Audio. The A udio E ngineering S ociety A step-by-step procedure for solving a mathematical problem. Ampere (I) The unit of electric current that is equal to one coulomb flowing per second. Also, I V R, wherein V Voltage in Volts, and R Resistance in Ohms (also, see Ohms Law). An electronic system which enables an input signal to control power from a source independent of the signal and thus be capable of delivering an output that bears some relationship to, and is generally greater than the input signal. An audio amplifier performs this function producing a relatively linear relationship between the input signal and the output signal. For more information on audio amplifiers, refer to Pre-Amplifier and Power Amplifier in the Glossary section of the Help File. The loudness (or intensity) of a sound at any given moment in time, which is represented on the vertical axis of the a u d i o workspace areas. Amplitude in audio terms is usually expressed in relative terms (the ratio of two levels) in dB (decibels), although sometimes it may be represented in absolute terms such as volts, or sound pressure level. An electronic system in which signals are represented, amplified, and processed utilizing continuous voltages andor currents (whose value could be expressed as an irrational number at any point in time) which are not quantized. DC MillenniumLIVE utilizes several digital simulations of analog systems in its algorithms. Solvents that are made up of cyclic hydrocarbons which, after they evaporate, tend to leave little or no residue on the surface on which they were used. Attenuation is the process of signal reduction, which is the opposite of the process of signal amplification. Most filters attenuate signals outside of their passband and feed signals through with no attenuation (or amplification) within their passband. Some filters, such as parametric and graphic equalizers are configured to provide either amplification or attenuation at any given frequency. Devices such as volume control potentiometers, L Pads, T Pads and H Pads are used to attenuate signals independent of frequency, i.e. flat. L Pads hold either the input or the output impedance constant as the attenuation factor is modified. T Pads hold both the input and the output impedance constant as the attenuation factor is changed. H Pads perform the same function as T Pads, only for balanced line systems. Here is a table of resistance multipliers for a symmetrical (equal input and output impedance) T Pad attenuator: R1 Attenuator Input Resistor R2 Attenuator Shunt Resistor R3 Attenuator Output Resistor To use this table, multiply the input (or output) impedance of your circuit by the numbers associated with the attenuation which you desire. Remember, this table of values requires that the input terminating impedance and the output terminating impedance of the circuits on each side of the attenuator be present and of the same value. To obtain values of attenuation which are not in this table, merely cascade quotTquot sections adding up to the value (in dB) which you desire. For example, to achieve 23 dB, cascade a 20 dB section with a 3 dB section. 1. Balanced quotXLRquot Standard: A. Pin 1 Shield (Common) C. Pin 3 - (Cold) 2. 14 inch Stereo Phone Plug (TRS) for Balanced Audio Circuits C. Sleeve Shield (Common) 3. 14 inch Mono Phone Plug (TR) for Unbalanced Audio Circuits B. Sleeve - (Shield) 4. RCA Phono Plug B. Sleeve - (Shield) 5. Amphenol 3 Pin Balanced Microphone Connector A. Pin 1 Shield (Common) 6. Amphenol 4 Pin Microphone Connector (Balanced and A. Pin 1 Shield (Common) B. Pin 2 (Hot) Unbalanced ( Note: Unbalanced output is with respect to Shield) C. Pin 3 (Hot) Balanced D. Pin 4 - (Cold) Balanced 7. DIN 5 Pin Connector (Tape Deck I O Connector) A. Pin 1 Right Channel Record Input B. Pin 2 Shield (Common) C. Pin 3 Right Channel Playback Output D. Pin 4 Left Channel Record Input E. Pin 5 Left Channel Playback Output 8. 1 8 inch Mono Phone Plug (TR) B. Sleeve - - (Shield) A. Tip Left Channel (Hot) B. Ring Right Channel - (Hot) C. Sleeve Shield (Common) 10. Modular Phone Jack (- 48 volt, 4 terminal , 2 line system United States) A. Red or Blue or Blue with White Stripe Line 1 B. Green or White or White with Blue Stripe Line 1 (Common) C. Yellow or Orange or Orange with White Stripe Line 2 - (Hot) D. Black or White or White with Orange Stripe Line 2 (Common) Denotes the most standard color code The range of frequencies between 20 Hz and 20 KHz. A very high quality audio system capable of reproducing this frequency range should be able to do so within - 3 dB. The following is a listing of some common audio sources and the portion of the audio spectrum which they typically occupy including their harmonics: When analog magnetic tapes are recorded or reproduced, the gap of the respective head (recording or playback) should ideally be perfectly normal (perpendicular) to the direction of the tape movement. If, in either of the two mentioned processes, the respective head gap is off-normal (off-azimuth,) two types of signal degradation will occur. The first phenomenon results in the loss of the high-end of the audio spectrum frequency response. The second effect produces a phase shifting of one channel with respect to the other, thereby smearing the stereophonic image. A similar phenomenon occurs when a monophonic half-track reel-to-reel tape is reproduced on a quarter track machine. Azimuth problems can be corrected by utilizing the Time Offset feature found in the File Converter, which is under the Filter Menu. Bandpass Filter A filter which only allows a range of frequencies to be passed without attenuation. A wide bandpass filter is one in which an upper and a lower corner frequency need to be defined, and often several octaves will be passed in between without attenuation. A narrow bandpass filter is one in which only a center frequency needs to be defined, and often has a bandwidth of an octave or less. The center frequency for a narrow bandpass filter is sometimes referred to as its resonant frequency. There are different bandpass shapes which can also be defined for narrow bandpass filters. Emile Berliner is widely know as the one who commercialized the lateral cut disc record format. He first introduced his products into the market place in 1895, although he had spent the previous 10-year period developing his product. However, Emile Berliner was not the inventor of the disc format or the lateral cut method for creating the undulations on a surface. These principles were outlined in the earlier Edison phonograph patents. quotBlastquot is a term which is used to describe a passage of sound on a recording which is disproportionately louder than the rest of the recording. quotBlastsquot can be created by poor instrument placement on acoustic recordings, poor mixes on electrical recordings, or by poor planning of microphone placement in live recordings. The term quotblastquot was used by recording engineers at least as early as the 1920s. The third (and final) generation of cylinder record which the Edison Company commercialized which was about 4 minutes in length. These records were made of a celluloid recording surfaced mounted on a plaster of Paris core. They were an improvement on the Edison Gold Molded black wax cylinders, which were only two minutes in length. The rotational speed for Blue Amberols (and black wax cylinders) is 160 RPM. A buffer is a memory sector which is used as a temporary storage location during input and output operations. The quotpreview bufferquot length is programmable in Diamond Cut. and is found in the preferences section of the Edit Menu. A Butterworth Filter produces a maximally flat amplitude characteristic in the pass band or the reject band (depending on whether it is used as a bandpass or a notch filter). It has a critically dampened response at the corner frequencies, having no ripple, and therefore it introduces little distortion into the signal which is feeding it. The Butterworth poles of signal transmittance are uniformly spaced on a semicircle, having its center on the imaginary axis. Its half-power frequencies are those at which the circle intersects the imaginary axis. Buzz usually refers to a series of harmonics related to the frequency of the AC power mains. It differs from Hum in sound, because it usually contains a large number of higher frequency harmonics. Buzz is best eliminated using the Harmonic Reject filter. An eight-bit word. Each sample of a monophonic wave file is represented by two eight-bit bytes. Two eight-bit bytes are used to represent all of the integer numbers between 0 to 65,535, which is the total dynamic range of your Diamond Cut Audio editor when it is operating in 16 bit mode. 1 kilobyte (Kbyte) 1,024 bytes Capacitance (C) Capacitance is the ratio of the electric charge given a body compared to the resultant change of potential. It is usually expressed in coulombs of charge per volt of potential change and its basic unit is the Farad. Energy is only stored (but not dissipated) in theoretical capacitance. Time constants for audio filters are created with a combination of resistors and capacitors in various configurations. High pass, low pass, bandpass, and notch filters can all be created with the appropriate combinations of resistors, capacitors, and operational amplifiers. The corner frequency for a simple first order RC filter 1 2 pi (R x C). The principle of capacitance (and conservation of charge) is involved in the operation of condenser and electret microphones and electrostatic loudspeakers and headphones. F (Farad) and uF (micro Farad or 1 X 10 -6 Farad) Note: pi 3.141592654 (approximately) Cassette Tape Equalization Time Constants Compact Cassette tapes (which operate at 1 78 ips) commonly utilize one of the following two equalization time constants based on the tape type: 1. Normal (IEC Type 1) (Usually Ferrous Oxide based): 2. High (IEC Type 2) Usually Chromium Oxide based): Classification of Amplifiers Audio Amplifiers can be broken down into several classifications based on their degree of conduction relative to its input signal. The VVA Virtual Valve Amplifier utilizes two of the following classifications. The others are included in the description for completeness: Class A: The device or devices conduct for a full 360 degrees of the input signal. These amplifiers can be wired either in single-ended or push-pull configurations. Class A Audio amplifiers are usually used in pre-amplifier stages, or low power amplifier applications. This circuit has the poorest electrical efficiency, but produces predominantly even order distortion. Class B: Two devices are operated out of phase with respect to one another. Each device conducts for only 180 degrees of the input signal. When the two amplified signals are combined, the full input waveform is represented, only amplified. This type of circuit is plagued by a phenomenon known as crossover distortion at low signal levels. This configuration is reserved for low performance PA amplifiers or AM (Amplitude Modulated) communications modulators. It is electrically efficient, but produces relatively large values of harmonic distortion especially at small signal levels. Class AB: Two devices are operated out of phase with respect to one another, just the same as the Class B configuration. However, each device conducts for more than 180 degrees of input signal, but less than 360 degrees. This configuration produces a reasonable tradeoff between electrical efficiency and low distortion. It is commonly found used in high high power audio power amplifiers. Since the circuit is symmetrical, distortion levels can be quite low. Class C: This configuration can consist of one or two devices which are conducting for anywhere between 90 to 180 degrees of the applied input signal. It is reserved for RF circuits only. Note: There are additional classifications of amplifiers involving tap switching, multiple rail, and pulse width modulation techniques, which have not been included in this listing. Clipping is a phenomenon, which occurs when a signal (or numerical value) exceeds a systems headroom. This concept applies to both analog and digital systems. The result of clipping is distortion. The amount of distortion produced depends on the amplitude of the over-driven signal. In Diamond Cut. clipping will occur anytime a signal or calculation produces a numerical value greater than 216 (or 65,536 counts or LSBs). Clipping can be observed as a flattening of the slope (horizontal line) of a signal at its peak on the Source or Destination workspace displays. Co-Axial Cable A coaxial cable is one constructed in a manner in which the signal conductor is located in the center of the return conductor with a dielectric located in-between. This provides three notable characteristics for the cable: 1. The center conductor is shielded from the effects of E fields which may be present. E field coupled current is returned back to signal ground with little effect on the signal itself. 2. The loop area formed between the two conductors is very small compared to other types of conductors thereby minimizing inductance and also susceptibility to H field coupling. 3. The cable exhibits a characteristic impedance which is independent of cable length (after past a few wavelengths) which is of a constant value related to its ratio of distributed inductance and capacitance. This makes the cable suitable for carrying RF (radio frequency) signals over long distances. Co-Axial cables are often used to carry low level signals from one audio device to another because of the first two mentioned characteristics. Comb Filter A comb filter (or Harmonic Reject filter) is a wave reject filter whose frequency rejection spectrum consists of a number of equi-spaced elements resembling the tines of a comb. This filter is useful for getting rid of Hum type noise containing more than just the line frequency fundamental component. In Diamond Cut. it is called the Harmonic Reject Filter, and for more details, please refer to the same. An electronic device which is used to reduce the dynamic range of an audio signal. They are often used to prevent overloading on certain mixer inputs (i.e. drums and vocals) in live performance applications. Radio stations often use them to make themselves quotsound louderquot when tuning across the radio band without violating any FCC regulations on maximum modulation or modulation index. The corner frequency of a filter is the frequency at which the signal has been attenuated by 3 dB relative to the pass band region of the filter. Crackle is a term used to describe relatively low levels of impulse noise found on old phonograph recordings. It is very similar to impulse noise, only the peak amplitude is much smaller in comparison. Crackle sort of sounds like Rice Krispies just after you pour the milk in the dish. Crackle is usually caused by slight imperfections in the record playing surface due to the use of coarse grain fillers in the record composition. Sometimes, crackle is caused by gas bubbles that occur in the surface as the record quotcuredquot after the stamping process. Crackle can be filtered out most effectively with the Impulse or Median Filter. Very old acoustic recordings may be even more effectively de-Crackled (and de-Hissed at the same time) with the Average Filter. dB (decibel) 110 of a bel. A bel is the basic unit for the measurement of sound intensity. It is a log scale measurement system used for relating the ratio of two acoustical or electrical parameters. Since electrical voltage, current, and power are used to represent sound through audio signals, the following mathematical relationships may be found to be useful when relating them in terms of outputs and inputs: dB (voltage) 20 log V output V input dB (current) 20 log I output I input dB (power) 10 log P output P input Note: A doubling of a voltage or current represents a 6 dB change. A doubling of power represents a 3 dB change. The following table shows the relationship between Voltage, Current, and Power ratios and Decibels: Current or Decibels Power Ratio Decibels Note: Standard Pitch is based on the tone A of 440 Hz. With this standard, the frequency of Middle C should actually be 261.626 Hz. NAB Equalization Curve (National Association of Broadcasters) The NAB Curve is a set of equalization frequency response contours which are used by manufacturers of analog tape recorders to compensate for the inductive nature of a tape head. The equalization time constants specified depend on tape speed. One pair of time constants are specified for 1 78 ips (inches per second) and 3 34 ips. Another pair of time constants are specified for 7 12 ips and 15 ips. The low frequency breakpoint for all speeds is 50 Hz. The high frequency breakpoint for 1 78 and 3 34 ips is specified as 1770 Hz. The high frequency breakpoint for 7 12 and 15 ips is specified as 3180 Hz. Unwanted disturbances superimposed upon a useful signal that tends to obscure its information content. Also, refer to Signal-to-Noise ratio for more information. Noise Gate A noise gate is an electronic device, which turns off a signal path when an input signal is below a predetermined threshold value. The Dynamics Processor produces a noise gate effect when you check the ExpanderGate function. You must set the ratio to the highest number for the best noise gate effect. A filter which attenuates all frequencies close to the center frequency of the filter setting. The degree of attenuation and the range of frequencies which are attenuated by this filter are determined by the filters Q or bandwidth. This type of filter is often used to minimize hum or acoustic feedback from a recording. This type of filter is sometimes referred to as a quotband reject filter.quot An octave is a group of eight musical notes and also a doubling of frequency. For example, the range of frequencies from 440 Hz to 880 Hz is 1 octave. The next octave will end at 1760 Hz. Note that in two octaves, the frequency has increased by a factor of four. A DC value of voltage or current added into a circuit to shift the quiescent operating point of a device or display. Offset is used in Diamond Cut to allow detail to be seen in a signal when the detail exists towards the top or bottom of the signal workspace display area. The unit of electrical resistance equal to the resistance of a circuit in which a potential difference of 1 Volt produces a current flow of 1 ampere. V I x R wherein V voltage in Volts, I current in Amperes, and R resistance (in Ohms) When an audio signal is applied to an audio device which is greater than the device can handle in a linear transfer manner, this creates a condition of quotover-modulation.quot It results in a distorted sound in the output of the device being over modulated. Sometimes, this condition is referred to as quotclipping,quot meaning that the amplification devices of an electronic system are either cutting-off or saturating due to overdrive. A variable electronic filter in which the following three parameters may be adjusted on each parametric channel: 2. Level (attenuation or amplification) Parametric equalizers are usually equipped with several parametric channels which can all be used simultaneously or each one can be individually bypassed. Pathe Freres Phonograph Company was a European based record and phonograph company, who utilized a somewhat unique groove modulation technique. Their method produced a vertical stylus displacement (like Edison Hill and Dale Diamond Discs and Cylinders) however this was accomplished by a different mechanism. The groove on these recordings is width modulated, and so when a conical stylus interacts with these groove width modulations, a vertical displacement is thereby produced. If you are transferring a Pathe 78 rpm recording with a stereophonic pickup cartridge, you will need to use the Diamond Cut Mono (L - R) file conversion algorithm. A Pentode is an electron tube (or valve) containing five elements. They include a cathode, anode, control grid, screen grid or beam deflector electrode, and a suppressor grid. They are most commonly used in audio power amplifiers, but are sometimes found in microphone pre-amplifiers. Typical beam power pentodes listed in ascending power levels include types 6BQ5EL84, 6L6GC, 5881, 7591, KT-66, 6CA7EL34, KT-88, and 6550. Phase Inversion Phase inversion is the phenomena when one of two signals has become 180 degrees phase shifted with respect to the other. This sometimes accidentally occurred on vinyl stereo recordings because the input leads to one of the two cutting lathe driver heads became swapped in location. This can be corrected by using the File Converter, using the Left or Right Phase-Invert feature. Pi (Greek Letter) is the symbol which relates the ratio of the circumference to the diameter of a circle. Pi C D wherein C Circumference of a Circle D Diameter of a Circle. Pi is approximately 3.141592654 Pink Noise is random noise, which is characterized as containing equal energy per unit octave. When viewed on an octave based spectrum analyzer, it will produce a flat horizontal line on the display. Pink Noise is useful for characterizing the frequency response of electronic systems and for analyzing room acoustic transmittance and resonance. Pink noise can be created through a two-step process using Diamond Cut. First, create white noise with the Makes Waves function. Next, process the signal through the Paragraphic equalizer using the factory preset labeled white to pink noise converter. Power is the time rate for the transfer of energy in any system. In other words, Power Energy time. In electrical terms, power is given in Watts and has the following relationships to Voltage, Current, and Resistance: P Power in Watts, V Voltage in Volts, and I Current in Amperes. R Resistance in ohms Power Amplifier (Power Amp) A power amplifier is a device that provides power amplification of an audio signal. Generally, this is the device that is used to drive a loudspeaker, the cutting head of a record lathe, or an audio transmission line, and is the final stage of amplification in an audio system. Audio power amplifiers generally develop somewhere between 10 to 1000 watts of output power, depending on make and model (although shake table audio amplifiers and AM radio transmitter modulators can be found which produce well over 50,000 watts). To minimize power loss in the transmission process, and to maximize system dampening factor, it is important to minimize voltage drops across loudspeaker distribution cables. Poor dampening factor will produce an ill-defined bottom-end (bass). Long distances between your power amplifier and your speaker system will require larger diameter cables. To determine the correct cable for your application, refer to the Wire Table provided in this Glossary. Pre-Amplifier (Pre-amp) A device that provides voltage amplification of an audio signal. Sometimes these devices also include equalization networks andor tone (bass, treble, loudness, etc.) controls. Pre-Emphasis The intentional added amplification which is sometimes applied to the top end of the audio spectrum during a recording or radio transmission process in order to raise the signal level at high frequencies substantially above the noise level of the system. This process is reversed during the reproduction process of the signal in order to recreate an overall flat frequency response. The result of this process is an improvement in the signal-to-noise ratio of the system. For example, the third specified time constant of 75 uSec associated with the RIAA equalization curve is pre-emphasis. Also, FM broadcast transmission utilizes a 75 uSec (or sometimes a 25 uSec) pre-emphasis to improve its signal-to-noise ratio. This process is reversed at your receiver (de-emphasis.) The Paragraphic equalizer contains 75uSec pre-emphasis and de-emphasis preset curves. Most of the filters and effects have a plethora of descriptive presets. Most often, the most efficient place to start when using a particular filter or effect would involve selecting one of the factory presets, and then tweaking the parameters to fine tune the system to your own personal taste. If you desire to keep a separate copy of your presets on diskette, it can be found in the Windows directory under DCArtpresets.ini Quiescent Point The Quiescent point (or operating point) of an amplification device like an electron tube or a transistor, refers to the bias established on its linear portion of the transfer function curve when the device is at rest (ie. no signal input applied). The Virtual Valve Amplifier allows you to adjust the Quiescent (operating) point of class A amplifiers anywhere from near cutoff to near saturation. RAM R andom A ccess M emory A digital electronic device for storing binary information temporarily. RAM performance is generally characterized in terms of its size in MBytes, and its access time in nanoseconds. Your computer will need a minimum of 8 MBytes of RAM to run the Diamond Cut application correctly. A system which can process a signal and output the signal at the same rate at which it is being fed into the system is said to be a real-time processor. The Diamond Cut algorithms can process signals in real-time or faster provided your platform is a 200 MHz Intel Pentium or higher. The exception to this rule is the 200 MHz Intel Pentium-Pro processor. Since it is not optimized for 16 bit applications, it cannot run all algorithms in real time or faster. A Real Time Analyzer is a form of spectrum analyzer used for the analysis of audio signals. Unlike conventional spectrum analyzers, it does not use a single filter in a scanning mode to produce an amplitude vs. frequency display, which is a relatively slow process. Instead, it processes audio signals in parallel, so that all frequency bands are displayed simultaneously. Generally, RTAs have 31 bands (in 1 3 octave increments) covering the frequency spectrum from 20 Hz to 20 KHz. They usually come with a calibrated electret microphone and a built-in pink noise generator for making acoustical measurements. A process wherein an alternating current signal is converted into direct current amplitude modulated envelope representation of the source. Often, some smoothing is applied to this signal with a set of time constants referred to as quotattackquot and quotdecay.quot This signal is used in such devices as dynamic filters, companders, compressors, expanders, spectral enhancers, and is digitally simulated in some of the Diamond Cut algorithms. The residue of a filtered signal is the algebraic difference between the filter output and its signal input. Diamond Cut allows you to hear the residue of two of its filters by enabling the Keep Residue function. The two filters that include this feature are the Continuous Noise Filter and the Harmonic Reject Filter. This feature has been included because in some cases, it may be useful as an aid to hear what you are filtering out of the signal source. This is particularly useful when adjusting the Harmonic Reject Filter when attempting to remove Hum or Buzz from a recording. Resistor Resistance (R) (Ohms) An basic electrical device which has electrical resistance, and is used to control the amount of current flow in a circuit. The unit of measurement for a resistor is the ohm. R E I wherein, R Resistance in ohms, E Voltage in Volts, and I Current in Amperes Standard RMA (Radio Manufacturers Association) Color Code: Color Significant Figure Decimal Multiplier Note: Actually, two pulses of light are produced per cycle of the line. But, for improved visibility, it is better to use every other pulse to light up the strobe. Note: The Diamond Cut program provides two bitmaps which you can download and use as phonograph strobes covering the important speeds. The following is a listing of some of the more common record types and the styli that they require: A. Modern LPs: 0.7 mil elliptical B. Early LPs: 1.5 mil truncated elliptical C. Transcription Recordings: 2.3 mil truncated elliptical D. Narrow Groove 78s such as Polydor: 2.4 mil truncated elliptical E. Late 1930s Lateral 78 RPM Discs: 2.8 mil truncated elliptical F. Standard Groove 78 RPM Discs: 3.0 mil truncated elliptical G. Pre-1935 Lateral Cut Electrical 78s: 3.3 mil truncated elliptical H. 1931 to 1935 RCA Pre-Grooved Home Recordings: 5.0 mil spherical I. Edison 80 RPM Diamond Discs: 3.7 mil spherical or non-truncated conical J. Edison Blue Amberol Cylinders: 3.7 to 4.2 mil non-truncated spherical K. Wide Groove Acoustical 78 Lateral Disc: 3.8 mil truncated elliptical L. Edison Wax Amberol Cylinders: 4.2 mil Spherical M. Edison White Wax, Brown Wax, Concert, and Gold Molded Cylinders: 7.4 mil Spherical N. Pathe 78s: 3.7 mil truncated conical O. Metal Stampers: Biradial of appropr iate dimensions P. Late 16 inch transcription discs: 2.0 mil truncated elliptical Q. Very early acoustical lateral cut discs: 4.0 truncated elliptical R. Etched-label Pathes up to 14 inches in diameter: 8.0 mil spherical S. Etched-label Pathes greater than 14 inches in diameter: 16.0 mil spherical T. Acetate and aluminum instantaneous discs: 6.0 mil elliptical or truncated elliptical Note: When stampers are played on a normal turntable equipped with a Biradial stylus, you will need to use the File Reversal feature so that it can be converted to forward play. Tape Recorder Speeds Time constants are exponential amplitude vs. time functions, which are realized with resistors and capacitors, or resistors and inductors. Tau R x C or Tau L R wherein Tau time constant in seconds, R resistance in Ohms, C capacitance in Farads, and L is inductance in Henries. The relationship between a simple first order filters corner frequency (F c ) and time constant is as follows: F c 1 (2 x pi x Tau) Note that the higher the value of time constant, the lower the corner frequency created. Some common time constants found in audio applications are as follows: 25 uSec Dolby based FM de-emphasis 70 uSec Type 1 (Normal Bias) Cassette Tape Eq 75 uSec Standard FM Broadcast de-emphasis 120 uSec Type 2 (High Bias) Cassette Tape Eq Additional audio time constants can be found under RIAA and NAB in this glossary. The instantaneous rate of change of a parameter (such as voltage amplitude or sound pressure level) with respect to time. (i.e. dV dt, dP dt, etc.) An alternating current device used to impedance match transducers and electronic circuits to one another. Sometimes, these devices are used with a unity turns ratio to provide isolation from one circuit to another rather than to impedance match the two. This is useful in audio applications when it is necessary to break a ground loop source of noise in a system. A Triode is an electron tube (or valve) containing three elements. They consist of an anode, cathode, and a control grid. Small changes in grid voltage produce large changes in values of current in the plate circuit (the ratio of delta plate current to delta grid voltage is its gain in transconductance or mu.) They are most commonly used in audio pre-amplifier, and other low-level applications. Typical triodes found in audio applications include the 12AX7 and 6SL7 high mu (gain), and the 12AU7 and 6SN7 medium mu devices. All of the devices listed are dual (two in one envelope). See Electron Tube. The frequency in a phonograph equalization curve below which the master was recorded with the cutting head operating in constant displacement mode rather than in constant velocity mode. This is used to limit the excursions of the cutting stylus so that bass notes do not cause the cutting stylus to break through to the adjoining groove wall. Here is a listing of the most common turnover frequencies utilized by brand and vintage: 200 Hz: Columbia (1925 - 1937) Victor (1925 - 1937) 250 Hz: Decca (1935 - 1949) 300 Hz: Columbia (1938 - End) 500 Hz: Brunswick Note 1: The temperature coefficient of resistance for copper wire 0.4 degree C Note 2: The resistance of a 2 conductor cable will be have to be doubled to account for the round trip. A slow periodic change in the pitch or low frequency flutter which may be present on phonograph, tape, or soundtrack recordings due to a non uniform velocity of the recording medium. Wow is generally a frequency modulating effect that occurs at a deviation rate between 0.5 to 6 Hz. The Wow could have been introduced in the recording process, the playback process, or a combination of both. Wow found on record recordings is usually caused by a non-concentric spindle hole. Wow found on tape recordings is generally caused by warped take-up or supply reels. Diamond Cut is not capable of correcting audio problems of this nature at this point in time. Wow and Flutter Wow and flutter is the combined FM effect of both mentioned parameters. The frequency spectrum in which this rate of frequency deviation is made is in the spectrum that exists between 0.5 to 250 Hz. This is the horizontal axis of a graph. In Diamond Cut. it contains the time information for your wave file that is divided up into ten equally spaced grids. This is the vertical axis of a graph. In Diamond Cut . it contains the amplitude information for your wave file that is divided up into four equally spaced grids.
Ta-ut-kontanter-forex
Js-moving-average