Dibuat oleh Olli Niemitalo pada tahun 2003-01-21, terakhir dimodifikasi 2012-08-04 Pada tahun 1998, saya memiliki beberapa waktu tambahan sementara yang lainnya membaca untuk ujian akhir di sekolah menengah atas, dan masuk ke pemrosesan sinyal digital. Saya menulis seperti yang saya pelajari, dan inilah hasilnya. Ini tidak sepenuhnya akurat di tempat tapi bisa menjadi tutorial bagus ke dunia audio DSP. Sebelumnya dokumen ini disebut tutorial pengolahan suara sonar untuk otak, tapi saya agak tumbuh dari identitas adegan saya selama bertahun-tahun. Nikmati seni ASCII Ini ditulis untuk penggemar pemrosesan sinyal digital audio (sesuai judulnya) dan yang lainnya membutuhkan informasi praktis mengenai subjek ini. Jika Anda tidak memilikinya sebagai pengalaman membaca linier dan menghadapi kesulitan, periksalah apakah ada sesuatu untuk membantu Anda dalam bab sebelumnya. Pada plot respons frekuensi filter, frekuensi linear dan skala magnitudo digunakan. Perubahan halaman dirancang untuk 60 printer linespage. Bab mengocok persamaan IIR ditulis oleh kakak ku Kalle. Dan, berkat Timo Tossavainen karena membagikan pengetahuan DSP-nya Copy dan gunakan teks ini dengan bebas. Oleh Olli Niemitalo, oiki. fi Perhatikan bahwa sampel dapat berarti (1) suara sampel atau (2) samplepoint Data suara sampel adalah setumpuk sampel, nilai amplitudo yang diambil dari gelombang suara sebenarnya. Tingkat pengambilan sampel adalah frekuensi tembakan. Misalnya, jika frekuensi 44100, 44100 sampel telah diambil dalam satu detik. Heres contoh sampling: Suara asli adalah kurva, dan 0s adalah titik sampel. Garis lurus horizontal adalah tingkat nol. Suara sampel hanya bisa mewakili frekuensi hingga setengah samplerate. Ini disebut frekuensi Nyquist. Bukti mudah: Anda harus menyimpan setidaknya dua samplepoints per siklus gelombang, bagian atas dan bawah gelombang untuk bisa merekonstruksinya nanti: Jika Anda mencoba memasukkan frekuensi Nyquist di atas suara sampel Anda, semua yang Anda dapatkan Adalah distorsi ekstra karena mereka muncul sebagai frekuensi yang lebih rendah. Suara terdiri dari komponen frekuensi. Mereka semua terlihat persis seperti gelombang sinus, tapi memiliki frekuensi, fase dan amplitudo yang berbeda. Mari kita lihat satu frekuensi: Sekarang, kita mengambil frekuensi yang sama dari suara lain dan perhatikan bahwa ia memiliki amplitudo yang sama, namun sebaliknya (diputar 180 derajat). Penggabungan dua sinyal dilakukan hanya dengan menambahkan keduanya. Jika kita melakukan hal yang sama dengan dua gelombang sinus ini, hasilnya akan menjadi: Diam. Jika kita memikirkan kasus lain, di mana perbedaan fasa kurang dari 180 derajat, kita mendapatkan gelombang sinus yang semuanya memiliki amplitudo dan fase yang berbeda, namun frekuensi yang sama. Heres cara untuk menghitung fase dan amplitudo sinewave yang dihasilkan. Mengkonversi amplitudo dan fasa menjadi satu bilangan kompleks, dimana sudut adalah fasa, dan nilai absolut amplitudo. Jika Anda melakukan ini pada kedua sinewave, Anda dapat menambahkannya sebagai bilangan kompleks. Seperti yang Anda lihat, fase gelombang sinus baru adalah 45 derajat dan amplitudo sqrt (1212) sqrt (2) sekitar 1.4 Sangat penting bahwa Anda memahami hal ini, karena dalam banyak kasus, lebih praktis untuk menyajikan amplitudo dan Fase frekuensi sebagai bilangan kompleks. Saat menambahkan dua suara sampel bersama-sama, Anda mungkin benar-benar menghapus beberapa frekuensi, yang memiliki fase berlawanan dan amplitudo yang sama. Amplitudo rata-rata suara yang dihasilkan adalah (untuk sumber asli independen) sqrt (a2b2) di mana a dan b adalah amplitudo sinyal asli. Penggunaan utama filter adalah untuk mengukur amplitudo komponen frekuensi dalam sebuah suara. Sebagai contoh, sebuah filter lowpass mematikan semua komponen frekuensi di atas frekuensi cutoff, dengan kata lain, mengalikan amplitudo dengan 0. Ini memungkinkan semua frekuensi di bawah frekuensi cutoff tidak diimbangi. Jika Anda menyelidiki perilaku filter lowpass dengan menggerakkan berbagai sinewave frekuensi yang berbeda melewatinya, dan mengukur amplifikasi, Anda akan mendapatkan respons frekuensi besarnya. Heres sebidang kurva respons frekuensi magnit dari filter lowpass: Frekuensi berada pada sumbu dan amplifikasi pada sumbu. Seperti yang Anda lihat, amplifikasi (penskalaan) frekuensi di bawah frekuensi cutoff adalah 1. Jadi, amplitudo mereka tidak terpengaruh dengan cara apa pun. Tapi amplitudo frekuensi di atas frekuensi cutoff bisa dikalikan nol sehingga lenyap. Filter tidak akan menambahkan komponen frekuensi baru ke suara. Mereka hanya bisa mengukur amplitudo frekuensi yang sudah ada. Misalnya, jika Anda memiliki sampel yang benar-benar sepi, Anda tidak bisa mengeluarkan suara darinya dengan menyaringnya. Juga, jika Anda memiliki sampel gelombang sinus dan menyaringnya, hasilnya akan tetap sama dengan gelombang sinus, hanya mungkin dengan amplitudo dan fasa yang berbeda - tidak ada frekuensi lain yang dapat muncul. Profesional tidak pernah bosan mengingatkan kita betapa pentingnya melupakan fase ini. Komponen frekuensi dalam suara memiliki amplitudo dan. Fase. Jika kita mengambil gelombang sinus dan gelombang kosinus, kita melihat keduanya mirip, namun memiliki perbedaan fasa pi2, seperempat dari siklus penuh. Juga, saat Anda memainkannya, mereka terdengar sama. Tapi, coba kenakan headset dan putar sinewave di saluran kiri dan gelombang kosinus pada saluran kanan. Sekarang Anda mendengar perbedaan Fase itu sendiri tidak mengandung informasi penting bagi kita sehingga tidak terdengar, namun perbedaan fasa, dari suatu frekuensi, antara kedua telinga dapat digunakan dalam memperkirakan posisi asal suara sehingga terdengar. Filter memiliki respons frekuensi yang besar, namun mereka juga memiliki respons frekuensi fase. Heres sebuah kurva contoh yang bisa dari filter lowpass: Jika Anda menyaring suara, nilai dari respons frekuensi fasa ditambahkan ke fase frekuensi suara asli. Linear (garis lurus) fase adalah hal yang sama seperti penundaan biasa, meskipun mungkin terlihat liar dalam plot jika terjadi beberapa kali. Jika Anda, misalnya, filter lowpass tidak memiliki respons frekuensi fase linier, Anda tidak dapat mengubahnya menjadi filter highpass dengan hanya mengurangkan outputnya dari yang asli dengan penundaan yang sama. Matematika kompleks dengan filter Respon filter untuk frekuensi tunggal dapat dinyatakan sebagai bilangan kompleks, di mana sudutnya adalah respons fase filter dan nilai mutlak respons besarnya. Bila Anda menerapkan filter ke suara, Anda benar-benar melakukan perkalian kompleks dari semua komponen frekuensi dalam suara berdasarkan nilai respons filter yang sesuai. (Baca bab Menambahkan dua sinewave bersama jika Anda merasa sulit untuk mengerti.) Contoh: Respon filter adalah (0,1) pada 1000Hz. Anda menyaring gelombang sinus, dengan informasi amplitudo fasa yang disajikan sebagai bilangan kompleks (0,1), dengan frekuensi yang sama dengannya: Fasa gelombang sinus diputar 90 derajat. Tidak ada perubahan amplitudo. Menggabungkan filter Respon gabungan kedua filter yang dimasukkan ke dalam serial ini adalah respons A yang dikalikan dengan respon B (bilangan kompleks seperti biasa). Jika Anda hanya perlu mengetahui respons besarnya, Anda juga bisa memperbanyak nilai absolut. Pada gambar tersebut, kedua filter mendapatkan masukan dari sumber yang sama. Keluaran mereka kemudian ditambahkan kembali, membentuk hasil akhir. Sekarang Anda perlu menggunakan tambahan dalam memecahkan respons gabungan. Filter FIR lebih lugas, dan lebih mudah dimengerti. Respons impuls yang terbatas berarti bahwa ketika input filter tetap nol untuk waktu tertentu, output filter juga menjadi nol. Filter respons impuls tak terbatas tidak pernah sepenuhnya mereda setelah mematikan masukan, namun lebih tenang dan lebih tenang sekalipun. Filter FIR dasar bisa jadi: di mana masukan berarti nilai sampel dimasukkan ke filter. Dalam kasus ini, orang akan berbicara tentang tap tap 3. Ini sampai ke koefisien (a0, a1, a2) apa filter ini akan lakukan terhadap suara. Memilih nilai koefisien adalah bagian yang paling sulit, dan kemudian bisa mendapatkannya nanti. Untuk merancang filter Anda sendiri, Anda perlu memahami beberapa matematika di belakang dan mengetahui metode yang benar. Pada contoh filter di atas, hanya nilai input terakhir yang digunakan. Dalam filter realtime, ini adalah persyaratan, karena Anda tidak tahu masukan masa depan. Dalam editor sampel dan semacamnya, Anda tidak memiliki batasan ini, karena Anda memiliki keseluruhan data masukan yang siap saat Anda mulai. Jika filter Anda adalah: dan Anda memerlukan versi realtime, cukup konversikan ke: Satu-satunya perbedaan adalah satu sampel penundaan pada filter realtime. Tidak seperti filter FIR, filter IIR juga menggunakan nilai keluaran sebelumnya dalam menciptakan keluaran mereka sekarang. Heres contoh sederhana: Ini bisa disebut 3 input, 3 output tap filter. Filter IIR tidak pernah bisa menggunakan nilai keluaran di masa depan, karena memang belum ada ada beberapa cara untuk menerapkan filter IIR yang sama. Beberapa mungkin lebih cepat daripada cara input-output-and-coefficient yang biasa. Bagaimanapun, setiap filter IIR dapat ditulis dalam bentuk ini, dan harus digunakan dalam desain filter dan perhitungan perhitungan. Respons impuls (Apa filter yang akan dilakukan terhadap impuls satu samplepoint) dari filter IIR sering terlihat kurang lebih seperti ini di sampledata: Beberapa filter IIR yang dirancang dengan buruk tidak stabil. Hal ini menyebabkan ouput menjadi lebih keras dan lebih keras daripada lebih tenang dan lebih tenang. Contoh sederhana dari hal ini adalah: output (t) input (t) 2output (t-1). Begitu mendapat data masukan, itu menjadi gila. Jenis filter yang dijelaskan di atas memproses sampel data dengan sampel. Tidak demikian, jika Anda menerapkan filter Anda menggunakan FFT, Fast Fourier Transformation. FFT biasanya beroperasi dengan potongan panjang 2n. Pertama, Anda harus menyiapkan respons impuls filter yang direncanakan. Kemudian konversikan, gunakan FFT, ke informasi spektral - bilangan kompleks yang mewakili fase dan amplitudo komponen frekuensi. Komponen ini disebut tempat sampah, karena frekuensi mereka tetap dan didistribusikan secara merata, dan jika data asli mengandung frekuensi di antara keduanya, maka sebagian besar energi frekuensi tersebut akan didistribusikan di antara tempat sampah di dekatnya. Sekarang, Anda juga FFT sampel data yang ingin Anda filter, dan kalikan frekuensi yang dihasilkan dengan filter dari filter. Kemudian IFFT (Inverse FFT) digunakan untuk mengubah informasi menjadi beberapa data sampel yang disaring. Jadi, perkalian dua data domain frekuensi menghasilkan konvolusi dari dua domain data waktu. Namun, ada tangkapan: FFT beroperasi dengan sinyal periodik, yaitu jika Anda memiliki respons impuls filter sepanjang potongan FFT maka data sampel non-nol di tengah potongan FFT akan menghasilkan pembengkokan buntut ekor. Dari filter di sekitar batas FFT. Untuk menghindari masalah ini, Anda dapat menggunakan FFT dua kali selama respons impuls filter, dan saat melakukan FFT pada data sampel, cukup mengisi buffer masukan FFT sampai setengah jalan dan tetapkan sisa masukan ke nol. Untuk input yang lebih lama, Anda akan memproses data dalam potongan seperti itu dan kemudian menambahkan potongan yang disaring yang dihasilkan bersamaan. Ini disebut metode overlap-add. Pilihan lainnya adalah overlap-save (lihat itu kalau mau). FFT juga dapat digunakan untuk menganalisis isi frekuensi data sampel, untuk alasan apapun. Jika Anda hanya mengambil sepotong data sampel, memiliki tepi tajam, yang buruk untuk FFT. Fungsi windowing digunakan untuk menghaluskan tepi ini. Dibesarkan kosinus, cos (x pi2) 2, adalah salah satu kemungkinan fungsi windowing. Di sini Anda melihat apa yang terjadi ketika Anda menerapkan fungsi windowing itu ke sejumlah data sampel: Terkadang (resampling, delay yang didefinisikan secara pasti) Anda perlu mendapatkan samplevalues dari antara samplepoint yang diketahui. Itu saat Anda membutuhkan interpolasi. Jika Anda tidak melakukan interpolasi, dan buang saja bagian pecahan sampleoffset Anda, Anda mendapatkan banyak distorsi frekuensi tinggi: Misalnya, samplepoints asli mencoba merepresentasikan gelombang sinus. Semakin dekat kurva interpolasi ke gelombang sinus, semakin baik algoritma interpolasi. Metode interpolasi simpliest adalah interpolasi linier. Garis lurus ditarik di antara dua samploint yang berdekatan: Masih terlihat cukup edgy untuk menjadi gelombang sinus. Namun, peningkatan yang tidak terpojok cukup signifikan. Ada juga kelemahan - frekuensi di bawah frekuensi Nyquist dilemahkan, bahkan lebih dari sekedar tanpa interpolasi. Heres rumus untuk interpolasi linier: baru tua (int) (lama (int1) - old (int)) fraksi, di mana int berarti bagian integer sampel offset dan fraksi bagian pecahan. Langkah selanjutnya adalah kurva Hermite, yang memberikan kualitas yang jauh lebih baik daripada interpolasi linier: Dengan interpolasi linier, Anda perlu mengetahui 2 samplepoint pada waktu untuk dapat menarik garis. Dengan kurva Hermite, jumlahnya adalah 4. Kurva interpolasi melewati dua titik tengah, dan titik 1 dan 4 digunakan untuk membentuk kurva. Rumusnya adalah kubik: Dan yang ini di sini adalah tempat a, b, c, d dipecahkan: Interpolasi yang sempurna juga ada. Dengan mengganti semua titik sampel dengan kurva sinc yang benar, sin (pi x) (pi x), dan dengan menambahkannya, Anda mendapatkan interpolasi yang tepat dan sempurna. Inilah salah satu samplepoint yang diganti dengan kurva sinc skala: Sinc curve tidak lama lagi, jadi Anda harus menggunakan semua samplepoints untuk menghitung satu nilai interpolasi. Solusi praktis adalah membatasi jumlah sampel untuk mengatakan 1000. Masih akan terlalu lambat untuk aplikasi realtime, namun akan memberikan ketepatan yang tinggi. Jika Anda bersikeras untuk menggunakan sinc dalam algoritma interpolasi realtime, cobalah menggunakan fungsi windowing dan sejumlah rendah (setidaknya 6) kurva sinc. Downsampling Jika Anda ingin melakukan downsample (mengurangi samplerate), Anda harus terlebih dahulu menyaring frekuensi Nyquist di atas, atau akan muncul sebagai distorsi pada sample downsampled. Dalam proses disain filter, Anda sering perlu melakukan kompromi. Untuk memiliki tepi tajam atau lereng curam dalam respons besarnya, Anda memerlukan filter yang besar, dan karena itu lambat. Dengan kata lain, filter dengan jumlah keran rendah praktis selalu memiliki respons magnitudo yang miring. Dalam kasus filter IIR, tepi tajam besarnya sering kali merupakan respons frekuensi fase yang jelek (sangat nonlinier), dan respons fase dekat-ke-linear merupakan respons magnitudo yang kecil. Dengan filter FIR, upaya untuk menciptakan tepi yang sangat tajam dapat menyebabkan melambai dalam besaran frekuensi terdekat. Filter IIR sangat bagus untuk rutinitas realtime, karena cepat, propertinya (misalnya frekuensi cutoff) dapat dengan cepat berubah di tengah tindakan, dan, suaranya terdengar seperti filter analog sebenarnya. ) Respon fase nonlinier dari filter IIR biasanya tidak penting. Filter FIR dapat digunakan di mana kualitas dan fase linier penting, misalnya, dalam contoh editor. Orang yang menyaring sinyal lain daripada suara, sering kali menginginkan respons frekuensi fase linier. Dengan sinyal stereo, penting untuk memiliki perubahan fasa yang identik pada saluran kiri dan kanan. Beberapa filter dan respons frekuensi magnitudo bergaya mereka: Jika Anda memiliki program perhitungan simbolis, saya sangat menyarankan Anda untuk menggunakannya dalam perhitungan mekanis, hanya untuk membuat hidup Anda lebih mudah. Turun adalah program DOS lama, namun tetap sangat berguna. Kebisingan putih Suara putih berarti jenis suara yang memiliki spektrum datar. Anda dapat dengan mudah membuatnya dengan menggunakan nomor acak sebagai samplevalues. Jika Anda ingin mengetahui respons frekuensi frekuensi dari sebuah filter, terapkan pada sampel panjang white noise dan kemudian jalankan analisis spektrum pada keluaran. Apa yang Anda lihat adalah besarnya respons frekuensi filter. Cara lain adalah dengan mengirim satu sampel impuls, yang awalnya memiliki spektrum datar. Sebuah impuls terlihat seperti ini di sampledata: 0, 0, 0, 0, 1, 0, 0, 0, 0 - dimana impulsnya adalah 1 di tengahnya. Dari keduanya, impuls itu lebih cepat, tapi menggunakan white noise bisa memberi hasil lebih bersih, karena kesalahan akan kurang terlihat. Untuk alasan yang sama, saat Anda menonton video, gambar diam akan terlihat lebih bersalju daripada gambar yang sedang berjalan. Mengambil analisis spektrum pada sampel yang panjang biasanya dilakukan dengan membaginya ke potongan yang lebih kecil, menganalisisnya secara terpisah dan kemudian mengambil rata-rata semua analisis. Pilihan pribadi saya disini akan menjadi program Cool Edit 96, yaitu untuk Windows. Metode kutub-nol adalah cara termudah merancang filter IIR cepat dan sederhana. Bila Anda telah mempelajarinya, Anda akan bisa merancang filter sendiri. Heres the complex Z-plane, yang digunakan dalam metode pole-zero: Bayangkan frekuensi yang akan dililitkan mengelilingi lingkaran unit. Pada sudut 0 kita memiliki 0Hz, pada pi2 kita memiliki samplerate4, pada pi kita memiliki samplerate2, frekuensi Nyquist. Anda tidak peduli dengan frekuensi yang lebih tinggi, karena tidak akan pernah muncul dalam sinyal, tapi bagaimanapun, pada 2pi (siklus penuh) kita memiliki frekuensi sampling. Jadi jika Anda menggunakan frekuensi sampling 44100 Hz, 0 Hz akan berada pada (1,0), 11025 Hz pada (0,1) dan 22050 Hz pada (-1,0). Apa itu kutub dan nol maka itu adalah hal-hal kecil yang lucu yang bisa Anda tempatkan di pesawat Z, seperti ini: Ada beberapa peraturan yang harus Anda ingat. Kutub harus selalu berada di dalam lingkaran unit, tidak pernah di luar atau di atasnya. Zeros bisa diletakkan dimana saja. Anda bisa menggunakan sejumlah kutub dan nol, tapi mereka semua memiliki pasangan konjugasi, jika tidak diposisikan di sumbu. Pasangan konjugasi berarti bahwa jika Anda memasukkan misalnya nol ke (0,6, 0,3), Anda harus memasukkan nol lain ke koordinat konjugasi, (0,6, -0,3). Dan hal yang sama dengan kutub. Tapi hei Apa yang kutub dan nol DO Polandia memperkuat frekuensi, nol menipis. Semakin dekat kutub adalah dengan frekuensi (pada lingkaran satuan, ingat), semakin diperkuat. Semakin dekat angka nol adalah frekuensi, semakin banyak yang dilemahkan. Sebuah nol pada lingkaran unit benar-benar membungkam frekuensi yang didudukinya. Sekarang bisa jadi saat yang tepat untuk mencobanya sendiri. Ada program disain filter gratis yang memungkinkan Anda bermain dengan kutub dan nol. Salah satu kandidatnya adalah: QEDesign 1000 demo for Windows. Di suatu tempat di Internet, Anda akan menemukannya. Merancang filter bandpass Filter simpliest yang dirancang menggunakan pole-zero adalah filter bandpass berikut: Poles memperkuat frekuensi, sehingga Anda bisa menarik kesimpulan bahwa frekuensi yang paling diperkuat adalah yang pada sudut yang sama dengan tiang. Dan Anda hampir benar Satu-satunya masalah berasal dari kutub konjugasi, yang juga memberi amplifikasi sendiri. Efeknya paling kuat pada sudut dekat dengan 0 dan pi, di mana jarak antara dua kutub adalah yang terkecil. Tapi jangan biarkan ini membingungkan Anda, segera kembali lagi nanti. Jadi sudut kutub menentukan frekuensi passband. Apa efek dari nilai absolut (radius) maka Seperti yang dinyatakan, kutub memperkuat frekuensi, dan amplifikasi lebih kuat saat tiang mendekati frekuensi. Pada filter bandpass kami, meningkatkan radius tiang menyebabkan respons besarnya menjadi lebih curam dan passband lebih sempit, seperti yang Anda lihat di sini: Posisi tiang: Plot respons frekuensi yang sesuai (dinormalisasi): Mari panggil radius r mulai sekarang. (Beberapa dari Anda mungkin ingat huruf q dari analog, filter resonan. Ini sama saja.) Dalam hal ini kita memiliki batasan: 0 r lt 1, karena kutub harus berada di dalam lingkaran satuan. Jadi perubahan r perubahan kecuraman, resonansi. Resonansi ini - bukan sihir, hanya satu frekuensi yang diperkuat lebih banyak daripada yang lain. Dari kutub dan nol untuk menyaring koefisien Ada fungsi transfer: di mana z adalah frekuensi, dalam bentuk koordinat lingkaran yang melingkar-sekitar-satuan-lingkaran. H (z) memberikan respon (kompleks) filter pada frekuensi z. P1, p2, p3 dan seterusnya adalah posisi kutub dan z1, z2, z3 dan seterusnya pada posisi nol. A0 adalah koefisien masukan pertama dari filter. Heres the IIR filter formula lagi, jika Anda lupa: Filter bandpass kami hanya memiliki satu tiang, dan pasangan konjugasinya, jadi kami dapat menyederhanakan fungsi transfer: dan ganti p1 dan p2 dengan koordinat kutub konjugasi: Mari memberi Pembagi melihat lebih dekat. Katakanlah: Kekuasaan z di sini sebenarnya adalah indeks pada output filter: Jadi kita tahu bagaimana menghitung koefisien sisi output dari posisi tiang: OK Katakanlah frekuensi passband berada pada bidang Z pada posisi ph: The Tiang berada pada sudut yang sama dengan frekuensi pada lingkaran satuan, namun memiliki jari-jari r. Oleh karena itu: Sekarang kita tahu bagaimana posisi tiang tergantung pada frekuensi, kita dapat menulis ulang koefisien sisi output: Tapi kita tidak boleh melupakan dividen (dari fungsi transfer), di mana kekuatan z adalah indeks ke input filter : Ini harus ditambahkan ke apa yang telah kita selesaikan dari sisi output: Selanjutnya kita harus memutuskan apa yang harus dimasukkan ke a0. Ini disebut normalisasi. Tujuan a0 adalah hanya untuk skala output dari filter. Dalam filter bandpass kami, kami menginginkan amplifikasi pada frekuensi passband menjadi 1. Jadi, kita dapat menulis persamaannya: Di sana sekarang siap saringan: Memperbaiki filter bandpass sederhana Kami dapat mengkompensasi efek tiang konjugasi dengan menambahkan Nol ke sumbu, di antara kutub. Misalnya, jika kita memiliki tiang pada koordinat (0,6, 0,5) dan (0,6, -0,5), nah beri angka nol pada (0,6, 0): Fungsi transfer untuk ini adalah: Koefisien sisi output sama persis dengan sebelumnya. . Koefisien sisi input dapat dipecahkan seperti ini: Jika Anda ingin menggunakan filter ini, Anda harus bisa melakukan normalisasi sendiri. Aku tidak akan melakukannya di sini. Kata-kata bijak Mudah membuat saringan lebih efisien: Ganda semua kutub dan nol. Respons frekuensi filter baru adalah kuadrat dari yang lama. Ada cara yang lebih baik, tapi ini yang paling mudah. Jika Anda menempatkan nol pada sebuah tiang, Anda menetralkan keduanya. Sebuah tiang di luar lingkaran unit menyebabkan filter menjadi tidak stabil. Sebuah tiang pada lingkaran unit dapat mengubah saringan menjadi osilator. Sejumlah besar kutub dan nol berarti sejumlah besar keran. Zeros mempengaruhi koefisien input, output poles. Kutub dan nol harus memiliki pasangan konjugasi, karena jika tidak, Anda akan mendapatkan koefisien filter yang kompleks dan, akibatnya, sinyal keluaran kompleks. Dengan nilai r rendah, frekuensi yang paling diperkuat tidak selalu pada sudut yang sama dengan tiang, karena efek tiang konjugasi. Cobalah untuk membedakan respons besarnya jika Anda menginginkan presisi yang tepat. Sebuah filter IIR tanpa tiang adalah filter FIR. 1 selalu berlaku. Bandpass dengan r Baca bab Desain filter IIR menggunakan metode pole-zero. Takik dengan r Semakin tinggi r, semakin sempit stopband. Lowpass dengan r Hal ini dapat dilakukan dengan beberapa cara: Semakin tinggi r, semakin kuat resonasinya. Filter lowpass resonan adalah tipe filter yang paling banyak digunakan dalam synthesizer. Allpass dengan r Highpass dengan r Impulse, sinc Jika Anda membaca tentang interpolasi sinc di bab Interpolasi suara sampel, Anda tahu bahwa Anda dapat mengganti satu puncak sampel (impuls) di sampledata dengan fungsi sinc yang diregangkan dengan benar. Benar diregangkan berarti amplitudesinc (t). Bila Anda menjalankan analisis spektrum pada dorongan hati, Anda mendapatkan spektrum datar dengan batas atas pada samplerate2, frekuensi Nyquist. Karena dorongan tulus, ini juga spektrum sinc: Anda bisa menarik kesimpulan bahwa Anda mendapatkan fungsi sinc jika Anda mengumpulkan semua frekuensi dari 0 sampai SR2, dan membagi jumlah dengan jumlah frekuensi, untuk memenuhi persamaan sinc (0) 1. Dan kamu benar. Dari analisis spektrum, Anda tahu bahwa semua frekuensi yang dijumlahkan sama memiliki amplitudo yang sama. Tapi bagaimana fasa mereka di pusat impuls Fungsi Sinc simetris sekitar x0, begitu juga kosinus - jadi sinc harus terbuat dari kosinus. Jika Anda menguji ini dengan sekitar 100 kosinus, Anda mendapatkan perkiraan yang cukup dekat dengan sinc. Jumlah dari semua frekuensi dari 0 sampai 1 (sebanding dengan SR2), dibagi dengan nomor mereka, dapat ditulis sebagai: (Ini berarti tidak terbatas) Seperti yang telah dilakukan di atas, x harus diganti dengan pi t, karena panjang siklus dosa adalah 2 pi, yang harus diregangkan ke 2 (yang merupakan panjang gelombang frekuensi Nyquist di sampledata). Fase shift Bagaimana jika kita mengganti kosinus dengan sine Mari kita mencobanya Theres formula universal (yang, btw, saya menemukan sendiri) yang bisa kita gunakan: Sekarang, jika kita mengganti semua dorongan dalam suara dengan fungsi baru ini, kita benar-benar melakukan sebuah -90 derajat pergeseran fasa Hal ini dapat dilakukan dengan membuat filter FIR, di mana koefisien diambil dari fungsi baru ini: (1-cos (pi t)) (pi t), namun dalam urutan terbalik, dengan mengganti t dengan - t , Jadi menjadi: (cos (pi t) -1) (pi t). Heres sebuah contoh yang menjelaskan mengapa hal itu perlu digunakan - bukan t: Katakanlah Anda ingin mengganti semua impuls dalam sinyal dengan urutan 1,2,3. Jika sinyal input adalah 0,1,0, akal sehat mengatakan seharusnya menjadi 1,2,3. Jika Anda hanya menggunakan 1,2,3 sebagai koefisien filter dalam urutan itu, sinyal yang disaring menjadi: Bukan yang Anda minta Tapi jika Anda menggunakan koefisien 3,2,1, Anda mendapatkan hasil yang benar, Ok, kembali ke -90 derajat pergeseran fasa filter. Bila Anda memilih koefisien filter dari (cos (pi t) -1) (pi t), pada t0 Anda dengan tidak benar mendapatkan pembagian nol. Hindari hal ini dengan menghitung batas t-gt0, di atas kertas, atau dengan proggy matematika. Jika Anda menggunakan otak Anda sedikit, Anda akan melihat itu adalah 0, karena rumus filter adalah jumlah sinus, dan sin (0) 0, jadi pada t0 itu adalah jumlah angka nol. Seperti sinc, fungsi baru kita tidak memiliki akhir, jadi kompromi harus dilakukan dalam jumlah keran. Hal ini menyebabkan gelombang dalam respon besarnya, dan atenuasi frekuensi yang paling rendah dan paling tinggi. Dengan menerapkan fungsi windowing ke koefisien, Anda bisa menyingkirkan ombak, tapi saya tidak tahu apapun yang bisa membantu redaman, kecuali keran lainnya. Fungsi windowing yang digunakan dengan FFT bekerja disini juga. Bagian tengah fungsi windowing harus berada pada t0, dan harus diregangkan sehingga ujungnya terbaring pada keran pertama dan terakhir. Anda juga bisa mendapatkan pergeseran fasa dari setiap sudut a: Perhatikan bahwa pembalikan t telah dilakukan di sini, jadi kita dapat mengambil koefisien langsung dari rumus ini. Batas t-gt0 secara alami cos (a), karena semua kosinus ditambahkan bersama-sama memiliki fase a pada x0. Jika Anda belum menyadarinya, ide utama pembuatan FIR filter adalah membuat fungsi yang berisi frekuensi yang ingin Anda lewati penyaringan. Amplitudo frekuensi pada fungsi secara langsung menentukan respons frekuensi besarnya filter. Tahapan frekuensi menentukan respons fase. Membalikkan koefisien hanya diperlukan dengan filter pergeseran fasa, karena filter yang tidak memperkenalkan pergeseran fasa apapun simetris di sekitar t0. Mendefinisikan rentang frekuensi yang disertakan Jika Anda menggunakan rumus sinc sebagai koefisien filter Anda, sebenarnya Anda tidak melakukan penyaringan, karena semua frekuensi dari 0 sampai Nyquist sama-sama disajikan dalam sinc. Disini Anda akan melihat bagaimana Anda dapat memilih frekuensi mana yang akan ada dalam rumus koefisien filter Anda. Ingat di mana kita awalnya mendapatkan sinc dari: Dalam integral, batas atas (1x) sebenarnya mewakili frekuensi tertinggi termasuk (1), dan batas bawah (0x) terendah (0). Jadi jika Anda menginginkan sebuah formula untuk filter bandpass, Anda dapat menulis: di mana bagian atas dan bawah adalah frekuensi cutoff sedemikian rupa sehingga 1 berarti frekuensi Nyquist, dan 0 berarti 0Hz. Sekarang taruh saja frekuensi yang Anda inginkan, hitung, dan ganti x dengan (pi t). Misalnya, jika Anda ingin membuat filter lowpass halfband (yang secara alami memiliki frekuensi cutoff pada samplerate4, sama seperti frekuensi Nyquist 2): Untuk membuat filter multi-band, Anda dapat menggabungkan beberapa formula filter bandpass dengan menambahkan Mereka bersama. Contoh equalizer Jika Anda ingin membuat equalizer (filter yang memungkinkan Anda menentukan besaran untuk frekuensi tertentu), Anda mungkin mengumpulkan banyak formula filter bandpass, yang diperkecil oleh besaran yang Anda inginkan untuk segmen frekuensi. Ini memberi Anda respons yang sangat besar yang terlihat sangat mirip dengan batu bata: Mungkin Anda ingin lebih mirip ini: Ada tiga cara. Cara pertama adalah dengan menggunakan batu bata yang lebih kecil, yang berarti bahwa Anda membagi frekuensi menjadi segmen yang lebih sempit dari sebelumnya dan menggunakan interpolasi untuk mendapatkan nilai besarnya untuk filter bandpass sempit baru yang kemudian Anda kombinasikan. Cara kedua adalah mendefinisikan polinomial (seperti ax3bx2cxd) yang memiliki karakteristik yang diinginkan (dan di mana x1 mewakili freqSR2), dan untuk membuat respons besarnya filter Anda mengikutinya. Ini mungkin. Cara ketiga adalah menambahkan beberapa formula bandwith ramp filter. Dalam respon besarnya solusi ini terlihat seperti garis lurus yang ditarik antara frekuensi yang didefinisikan di dekatnya. Ini juga mungkin, dan, menurut pendapat saya, solusi terbaik. Respon frekuensi magnitudo berbentuk polinomial Dalam sinc, semua gelombang kosinus ditambahkan bersama memiliki amplitudo yang sama, seperti yang Anda lihat di sini - semua frekuensi diperlakukan sama: Anda dapat mengubahnya dengan meletakkan di sana fungsi g () yang mendefinisikan amplitudo gelombang kosinus Dari frekuensi yang berbeda: Jika fungsi g (x) adalah bentuk axb, perhitungannya seperti ini: Sebagai contoh sederhana, jika kita menginginkan respons frekuensi besarnya menjadi garis lurus, mulai dari 0 pada 0Hz, dan berakhir pada 1 pada SR2, kita definisikan g (x) x: Dan perhitungan rumus koefisien filter untuk ini: Dalam kasus lain, untuk mendapatkan formula untuk polinomial penuh, lakukan perhitungan untuk masing-masing persyaratannya (axb) secara terpisah dan jumlah hasilnya. Bandpass magnitude-ramp Heres contoh respon frekuensi besarnya dari filter jalan: Untuk membuat jalan bandpass, Anda harus terlebih dahulu menentukan polinomial g (x) yang menjelaskan bagaimana besarnya berperilaku di dalam batas bandpass. Besarnya linier di dalam batas, sehingga polinomial g (x) harus berbentuk cxd. C dan d dapat dipecahkan dari persamaan: di mana x1 adalah batas frekuensi yang lebih rendah, dan x2 lebih tinggi. Y1 dan y2 adalah besaran dari frekuensi batas. Ingat bahwa disini x1 sama dengan frequencySR2. OK, berikut ini adalah penyelesaian c dan d: G (x) cxd adalah polinomial, dan Anda sudah tahu bagaimana membuat respons frekuensi besarnya memiliki bentuk yang sama (Respon frekuensi berbentuk polinomial berukuran besar) sebagai polinomial. Anda juga sudah tahu bagaimana memasukkan hanya rentang frekuensi yang terbatas (Bagian Mendefinisikan rentang frekuensi yang disertakan) dalam rumus koefisien Anda. Kombinasikan pengetahuan ini, dan Anda bisa menulis rumus koefisien untuk ramp bandpass filter: Catatan tentang penerapan equalizer. Jika equalizer harus disesuaikan realtime, menghitung ulang seluruh rumus filter equalizer dengan semua fungsi trigonometri dapat berubah terlalu berat. Mungkin lebih baik untuk mendahului koefisien untuk beberapa filter yang saling tumpang tindih, misalnya untuk equalizer tiga saluran: Ketika menghitung koefisien untuk keseluruhan equalizer, cukup pilih koefisien yang sesuai dari ini, skala sesuai dengan slider equalizer, dan penjumlahan. Jika Anda mengambil koefisien filter FIR langsung dari formula filter Anda, Anda mendapatkan respons magnit berganda. Alasannya sederhana: Jumlah koefisien terbatas, namun rumus filternya tidak, namun terus memiliki nilai nol di luar kisaran yang Anda gunakan untuk koefisien. Fungsi windowing membantu. Not using a windowing function is the same thing as using a rectangular ( flat inside its limits) windowing function. Using a windowing function means that you multiply the values taken from your infinitely long filter formula by the corresponding values taken from your finitely long windowing function, and use the results as filter coefficients. Here are some windowing functions, and the produced magnitude responses of a FIR lowpass filter with a low number of taps, illustrated: As you see, the steeper the cutoff, the more waves you get. Also, if wed look at the magnitude responses in dB scale, wed notice that from the three, cos4 gives the best stopband ( the frequency range that should have 0 magnitude) attenuation. Mathematically, multiplication in the time domain is convolution in the frequency domain, and windowing is exactly that. (Also, multiplication in the frequency domain is convolution in the time domain.) I hope i didnt slam too many new words to your face. Time domain means the familiar time-amplitude world, where we do all the FIR and IIR filtering. The frequency domain means the frequency-amplitudeampphase world that you get into through Fourier transformation. And convolution In the time domain, FIR filtering is convolution of the input signal with the filter coefficients. Say you convolute 0,1,0,0,2,0,1,0 with 1,2,3 (where 2 is at the center): Youll get 1,2,3,2,4,7,2,3. If you understand this example, you surely understand convolution too. Ideally (impossible), there would be no windowing, just the constant value 1 infinitely in time. And a steady constant value in the time domain is same as 0Hz in the frequency domain, and if you (in the frequency domain) convolute with 0Hz, it is the same as no convolution. Convolution in the frequency domain equals to multiplication in the time domain, and convolution in the time domain equals to multiplication in the frequency domain. Sounds simple, eh But note that in this frequency domain, there are positive AND NEGATIVE frequencies. Youll learn about those in chapter Positive and negative frequencies. Words of wisdom You get flat (but not necessarily continuous) phase response if your filter (filter coefficients) is symmetrical or antisymmetrical (sides are symmetrical but have opposite signs, and the center crosses zero) around t0, even if you limit the number of coefs andor window them. Sometimes you can optimize your filter code a lot. Some coefficients may turn zero, so you can skip their multiplications. If your filter is symmetrical around t0, you can instead of input(t)ainput(-t)a write (input(t)input(-t))a). If your filter is antisymmetrical around t0, replace input(t)a-input(-t)a) with (input(t)-input(-t))a. Sinc(t) is 1 at t0, and 0 at other integer t values. Calculating the limit t-gt0 is very simple. If your filter formula was originally a sum cosines (meaning its not a phase shift filter), the limit t-gt0 is simply the area of the magnitude frequency response, in such way that the area of no filtering is 1. The actual filter implementation (after possible coefficient calculations) depends much on how the input data is fed to the filter. I can see three cases: You have the whole input data in front of you right when you start. A sample editor is a good example on this. This is the easiest case. With FIR filters, just take values from the input data, multiply with coefficients and sum, like this: output(t) a0input(t-2) a1input(t-1) a2input(t) a3input(t1) a4input(t2). The only problem is what to do at the start and at the end of the input table, because reading data from outside it would only cause problems and mispredictability. A lazy but well working solution is to pad the input data with zeroes, like this: This is how its mostly done with FFT filtering. With FIR filters, it isnt that hard to write a version of the routine that only uses a limited range of its taps, like this: and to use that version at the start and at the end. For this, it is easiest if you have a table of coefficients instead of hard-coding them into the routine. Data is fed to the filter in small chunks, but it is continuous over the chunk borders. This is the most common situation in programs handling realtime audio. One sample at a time. Case 2 can be treated as this, because the chunks can always be chopped into single samples. It is a fact that you cannot use future inputs in this case, so a FIR filter would have to be of form such as: output(t) a0input(t-4) a1input(t-3) a2input(t-2) a3input(t-1) a4input(t). Clearly this kind of a filter creates a delay, but thats just a thing you have to learn to live with. Also, you only get in one sample at a time, which is not enough for filtering, so you have to store the old input values somehow. This is done using a circular buffer. The buffer is circular, because otherwise youd soon run out of memory. Heres a set of pictures to explain the scheme: The buffer must be at least as long as the filter, but it is practical to set the length to an integer power of 2 (In the above example: 2532), because then you can use the binary AND operation to handle pointer wrapping always after increasing or decreasing one (In the above example, AND with 31). Even better, use byte or word instructions, and wrapping will be automatically handled in overunderflows caused by the natural limits of byte or word. Note that the buffer should be filled with zeroes before starting. A similar circular buffer scheme is also often the best solution for implementing the output part of an IIR filter, no matter how the input part was realized. There are both positive and negative frequencies. Until now we havent had to know this, because we have been able to do all the calculations by using sines as frequencies. Dont be fooled that positive frequencies would be sines, and negative ones something else, because that is not the case. In all real (meaning, not complex) signals, positive and negative frequencies are equal, whereas in a complex signal the positive and negative frequencies dont depend on each other. A single sinewave (real) consists of a positive and a negative frequency. So any sine frequency could be expressed as a sum of its positive and negative component. A single, positive or negative, frequency is: and could also be written as: As stated, a sinewave consists of a positive and a negative frequency component. Heres the proof: (The phase of the negative frequency must also be inverted, because it rotates to the other direction) As you see, the imaginary parts nullify each others, and all that remains is the real part, the sine wave. Amplitude of the sine wave is the sum of the amplitudes of the positive and the negative frequency component (which are the same). This also proves that in any real signal, positive and negative frequencies are equal, because a real signal can be constructed of sine waves. The complex Z-plane is a good place to look at positive and negative frequencies: Positive frequencies are on the upper half of the circle and negative frequencies on the lower half. They meet at angles 0 and the Nyquist frequency. Aliasing usually means that when you try to create a sine wave of a frequency greater than the Nyquist frequency, you get another frequency below the Nyquist frequency as result. The new frequency looks like as if the original frequency would have reflected around the Nyquist frequency. Heres an example: The cause of aliasing can be easily explained with positive and negative frequencies. The positive component of the sine wave actually gets over the Nyquist frequency, but as it follows the unit circle, it ends up on the side of negative frequencies And, for the same reasons, the negative component arrives on the side of positive frequencies: The result is a sine wave, of frequency SR-f. Analytic signal It is sometimes needed to first create a version of the original signal that only contains the positive frequencies. A signal like that is called an analytic signal, and it is complex. How does one get rid of the negative frequencies Through filtering It is possible to do the job with an IIR filter that doesnt follow the conjugate-pair-poles-and-zeros rule, but a FIR filter is significantly easier to create. Well use the old formula that we first used to create sinc: but this time, instead of cosines, only including the positive frequencies: As you see, the filter coefficients are complex. We should also halve the amplitude of the positive frequency (it should be half of the amplitude of the cosine, because the negative component is gone) but thats not necessary, because itd only scale the magnitude. To convert the complex analytic signal back to real, just throw away the imaginary parts and all the frequencies get a conjugate (on the z-plane) pair frequency. Here the amplitudes drop to half, but as we skipped the halving in the filtering phase, it is only welcome. The real to analytic signal conversion could also be a good spot for filtering the signal in other ways, because you can combine other filters with the negative frequency removal filter. Amplitude modulation Amplitude modulation means multiplying two signals. All samplepoints in the modulated signal are multiplied by the corresponding samplepoints in the modulator signal. Heres an example: What happens if we modulate a signal with a sinewave The original signal is (as we have learned) a sum of frequecy components, sinewaves of various frequencies, amplitudes and phases. Note that the signal we are talking about here is real, not complex. Say sNUMBER is one of the frequency components. So, we can write the original signal as: Now, if we multiply this signal with the modulator signal m, we get: This is good, because as you see, its the same as if the frequency components were processed separately, so we can also look at what happens to each frequency component separately. A frequency component can be written as: where amp is the amplitude, f the frequency and a the phase. The modulator sine can be written the same way (Only added the letter m): Multiply those and you get: If we discard the phase and amplitude information, we get: which is two frequencies instead of the origial one. Heres a graph that shows how the frequencies get shifted and copied. The original frequency is on the - axis and the resulting frequencyfrequencies on the axis: In the graph Modulated, the frequencies that would seem to go below zero, get aliased and therefore reflect back to above zero. In sampled signal, the Nyquist frequency also mirrors the frequencies. Frequency shifting With some tweaking and limitations, you could make a frequency shifter by using sinewave modulation, but theres a better way. Lets try modulating the signal with e(i mf x) instead of cos(mf x). Phases and amplitudes are irrelevant, so ive just ignored them. (I hope you dont mind) Lets see what happens to a single positivenegative frequency when it is modulated: The answer is very simple. The original frequency got shifted by the modulator frequency. Notice how the rule Multiplication in the time domain is convolution in the frequency domain. applies here also. Heres an example on the z-plane unit circle. p0, p1, p2 are the positive frequencies and n0, n1, n2 their negative conjugate frequencies. Say the modulator frequency rotates the frequencies 14 full cycle counterclockwise: In the modulated signal, the original pair frequencies (like p0 and n0) are no longer conjugate pairs. Thats bad. Another bad thing is that negative frequencies get on the side of positive frequencies and vice versa. But if we first filter all the negative, and those of the positive frequencies that would arrive on the wrong side of the cirle, and then modulate the filtered signal: (The filter formula is in the chapter A collection of FIR filters in section Combined negative frequency removal and bandpass) Now it looks better To make this filtered amp modulated complex signal back to real again, just discard the imaginary part and all the frequencies get a conjugate pair: For most sounds, frequency shifing doesnt do a very good job, because they consist of a fundamental frequency and its harmonics. Harmonic frequencies are integer multiples of the fundamental frequency. After you have shift all these frequencies by the same constant frequency, they no longer are harmonics of the fundamental frequency. There are ways to do scaling instead of shifting, but just scaling the frequencies would be same as resampling, and resampling also stretches the sound in time, so it has to be something smarter. The main idea is to divide the sound into narrow frequency bands and to shiftscale them separately. OK, so frequencies usually come with harmonics - Why Just think where sounds in nature originate from: vocal cords in our throat, quitar strings, air inside a flute. All vibrating objects, and you have probably learned at school that objects have several frequencies in which they like to vibrate, and those frequencies are harmonics of some frequency. What happens in those objects is that they get energy from somewhere (moving air, players fingers, air turbulence), which starts all kinds of vibrationsfrequencies to travel in them. When the frequencies get reflected, or say, go around a church bell, they meet other copies of themselves. If the copies are in the same phase when they meet, they amplify each other. In the opposite phases they attenuate each other. Soon, only few frequencies remain, and these frequencies are all harmonics of same frequency. Like so often in physics, this is just a simplified model. A note about notation. ) The fundamental frequency itself is called the 1st harmonic, fundamental2 the 2nd, fundamental3 the 3rd, and so on. Chromatic scale In music, harmonics play a very important role. The chromatic scale, used in most western music, is divided into octaves, and each octave is divided into 12 notes. The step between two adjanced notes is called a halftone. A halftone is divided into hundred cents. An octave up (12 halftones) means doubling the frequency, an octave down (-12 halftones) means halving it. If we look at all the notes defined in the chromatic scale on a logarithmic frequency scale, we note that they are evenly located. This means that the ratio between the frequencies of any two adjacent notes is a constant. The definition of octave causes that constant12 2, so constant 2(112) 1.059463. If you know the frequency of a note and want the frequency of the note n halftones up (Use negative n to go downwards) from it, the new frequency is 2(n12) times the old frequency. If you want to go n octaves up, multiply by 2n. But why 12 notes per octave As said, harmonics are important, so it would be a good thing to have a scale where you can form harmonics. Lets see how well the chromatic scale can represent harmonics. The first harmonic is at the note itself: 0 halfnotes 1. The second harmonic is at 1 octave 2. The third harmonic is very close to 1 octave 7 halftones 19 halftones 2(1912) 2.996614. Dan seterusnya. Heres a table that shows how and how well harmonics can be constructed: Not bad at all The lowest harmonics are the most important, and as you see, the errors with them are tiny. I also tried this with other numbers than 12, but 12 was clearly the best of those below 30. So, the ancient Chinese did a very good choice The above table could also be used as reference when tuning an instrument, for example a piano (bad example - no digital tuning in pianos), to play some keys and chords more beautifully, by forcing some notes to be exact harmonics of some other notes. A common agreement is that one of the notes, middle-a, is defined to be at 440Hz. This is just to ensure that different instruments are in tune. Flanger is simply: where d is the length of the variable delay. D values have a lower limit, and the variation comes from sine: Because d is not integer, we must interpolate. Most probably, annoying high frequency hissing still appears. It can be reduced by lowpass filtering the delayed signal. Wavetable synthesis means that the instruments being played are constructed of sampled sound data. MOD music is a well-known example. Also most of the basic home synthesizers use wavetable synthesis. Say you have a sampled instrument, and want to play it at frequency f 440Hz, which is middle A in the chromatic scale. To be able to do this, you need to know A) the samplerate of the sample and the frequency of the sampled instrument, or B) the wavelength of the instrument expressed as number of samples (doesnt have to be integer). So you decide to precalculate the wavelength to speed up the realtime routines a little: The samplerate of your mixing system, SR, is 44100Hz. Now that you know this, you can calculate the new wavelength, the one you want (number of samples): In the mixer innerloop, a sample offset variable is used in pointing to the sampledata. Every time a value is read from the sampledata and output for further mixing, sample offset is advanced by adding variable A to it. Now we must define A so that ol (256) is stretched (here shortened) to nl (100.22727), in other words, so that for ol samplepoints in the sampledata, you produce nl output values: Everything on one line: Thats it By using A as the addvalue, you get the right tone. Click removal There are some situations when unwanted clicks appear in the output sound of a simple wavetable synthesizer: Abrupt volume (or panningbalance) changes. A sample starts to play and it doesnt start from zero amplitude. A sample is played to the end and it doesnt end at zero amplitude. (Biased sampledata or badly cut out sample) A sample is killed abruptly, mostly happens when new notes kill the old ones. Poor loops in a sample. And what does help Heres some advice: Volume changes must be smoothed, maybe ramped, so that itll always take a short time for the new volume to replace the old. Clicky sample starts can be muffled, meaning that the volume is first set to zero and then slided up. This could of course be done beforehand too, and some think muffling sample starts is wrong, because the click may be deliberate. Some drum sounds lose a lot of their power when the starts are muffled. Another case is when the playing of a sample is not started from its beginning. That will most probably cause a click, but muffling is not the only aid - starting to play from the nearest zero crossing also helps. Abrupt sample ends should also be faded down. This may require some sort of prediction, if you want to fade down the sound before its ran over by another sound. This prediction can be made by using a short information delay buffer. It may be easier to just use more channels, to allow the new sound to start while the other one is being faded out in the background, on another channel. When the sampledata ends at a value other than zero, the cause may be that the sampledata is not centered around the zero level, or that the creator of the sample has just cut the end of the sample away. The easiest way to fix this is to fade out the end of the sample beforehand. However, this is not always possible. Symmetric form Turning an IIR filter backwards Getting rid of output(tn) Getting rid of input(tn) FIR frequency response IIR frequency response. Olli wrote he tried to make his text as down-to-earth as possible. Well, heres a more mathematical approach. But Ive still tried to make this intuitive and FUN rather than boring myself with lengthy proofs. This also means that there may be errors, most probably in signs. Symmetric form Say you have this IIR filter: You can put its equation to this symmetric form: Now define a new function, middle(t): You can rewrite this as: Notice how the transition from input(t) to middle(t) is a FIR filter and the transition from output(t) to middle(t) is another. So the IIR filter in fact consists of two FIR filters facing each other. This gives a simple approach to frequency response calculations (see the section IIR frequency response). Turning an IIR filter backwards You can solve input(t) from the IIR equation: Now swap input and output and you have a filter that undoes what the original did. But if the frequency response of the original filter was ZERO for some frequency, the inverted one will amplify that frequency INFINITELY. This is just logical. The inverted filter will also have an opposite phase shift, so that if R(f) is the frequency response of the original filter as a complex number and r(f) is the frequency response of the inverted filter, R(f)r(f)1 for every f. Getting rid of output(tn) Say you have somehow found that you need an IIR filter like this: You need to know both output(t2) and output(t-2) to be able to compute output(t). Doesnt seem very practical. But you can shuffle the equation a little: Now define a new variable ut2 and use it instead of t: Then solve output(u): Now you can use the filter. Getting rid of input(tn) Notice how in the previous example, input(t) became input(u-2). Had there been input(t1), it would have become input(u-1) which can be used in real time filters. Generally, you can get rid of input(tn) this way if the equation also uses output(tm) where mn, because you can define utm which turns input(tn) to input(u-(m-n)) which you get in time. If mltn, this is not possible: Here m0 and n1, so you cant get rid of input(t1) and keep the filter mathematically equivalent to the original. However, you can delay the output by one time unit: Usually, this small delay doesnt matter. But it changes the phase frequency response of the filter and this DOES matter if you then mix the filtered signal with the original one or others derived from it in that case, youd better make sure that all of the signals have the same delay. (Except if you happen to like the extra effect.) (For example, if you have a filter output(t)input(t-1), it doesnt do much as such. But if you mix the filtered signal with the original one, the mixing becomes a filter in itself and you can compute its frequency response and all.) If you try to force the original filter through the utm trick by introducing a dummy 0output(t1) term: youll just get division by zero. FIR frequency response Treat a sine wave as a rotating phasor e(it2piffs) where: The real component of this phasor is the regular sine wave. The neat thing about this is that you can multiply it with various complex numbers to scale the magnitude and shift the phase at the same time. By defining ze(i2piffs), the phasor can be written as zt. This is the same z that is used in pole-zero calculations (see chapter IIR filter design using pole-zero method). Heres the general FIR equation: Now, lets look what the filter does to an infinitely long sine wave with frequency f. But this sine wave can be replaced with the rotating phasor if we then throw away the imaginary component of the output. m(k) is real so the real and imaginary components cant affect each other. Here the zt factor doesnt depend on k, so it can be moved outside the sum: z depends on f (ze(i2piffs), remember) but the value of the sum doesnt depend on t. Ill call it R(f): output(t) is a rotating phasor at the same frequency as input(t) it just has a different amplitude and phase as defined by R(f). This means that for an infinitely long sine wave of frequency f, R(f) shows how the filter affects its amplitude and phase. In other words, R(f) is the frequency response of the filter. Its a complex function. If you dont remember what this means, see section Complex math with filters in chapter Whats a filter in this file. IIR frequency response When two filters are concatenated so that one filters output is fed to the other filters input, the responses are multiplied at each frequency: A filter that just connects its input to its output and doesnt change the signal at all has a frequency response of 1 at all frequencies: Now assume that we have a filter with frequency response R(f) and we make another filter with frequency response Rinv(f) that UNDOES everything the first filter did to the signal when they are concatenated. So the inverse filter also has an inverse frequency response. Remember, an IIR filter consists of two FIR filters facing each other (see section Symmetric form). This setup can be treated as a normal FIR filter followed by an inverted FIR filter: This means that if you can calculate the frequency responses of the two FIR filters, you can calculate the IIR frequency response by dividing one with the other. An example. You have this IIR filter. Change the names of functions a little: Compute the frequency response of filter input1-gtoutput1 (originally input-gtmiddle). The general formulas: In this particular case: The input2-gtoutput2 (originally output-gtmiddle) filter: Now the whole IIR: To actually calculate the frequency response at some frequency, youd apply Eulers formula and the usual complex number rules: R in the filters means resonance, steepness and narrowness. Fastest and simplest lowpass ever Fast lowpass with resonance v1 19 Comments raquo Thanks for posting this. It8217s a nice collection of audio DSP nuggets. May I suggest that the URL at the top of the original text document (iki. fiodspdspstuff. txt ) be pointed directly to this page. Comment by ColdCold 8212 2009-11-16 16:06 Thanks Mate, Greatly appreciate this tutorial. DSP in simple terms is not easy to come by on the Web Comment by Don 8212 2010-05-10 04:29 Thanks a lot. Very useful concepts explained in a lucid manner. Comment by Ravi 8212 2010-08-30 14:59 Hi, About notch filter.. Why I can8217t get the frequency cut effect Sample rate:1600 freq 1950 q 0.1 z1x cos(2pifreqsamplerate) a0a2 ((1-q)(1-q))(2(fabs(z1x)1)) q a1 -2z1xa0a2 b1 2z1xq b2 -(qq) 8212821282128212- frequency: 1950.000000 q: 0.100000 z1x: 0.195090 a0a2: 0.438887 a1: -0.171245 b1: 0.039018 b2: -0.010000 Each sample calculation: 82128212821282128212821282128212821282128212821282128212 reg0 a0a2 ((double)samplecurrentsampleminus2) a1sampleminus1 b1reg1 b2reg2 reg2 reg1 reg1 reg0 82128212821282128212821282128212821282128212821282128212 Is it correct Output is clean voice, but 1950Hz carrier is still there. BR Comment by Alexander Vangelov 8212 2011-03-16 22:46 Freq should be between 0 and samplerate2. (Just a quick comment before I go to bed) Comment by Olli Niemitalo 8212 2011-03-17 00:53 Thank you, it works :) I missed a zerro digit in parametters (just before I go to bed) Sample rate: 16000 Freq: 1950.000000 q: 0.400000 z1x: 0.720854 a0a2: 0.504599 a1: -0.727484 b1: -0.576683 b2: 0.160000 Comment by Alexander Vangelov 8212 2011-03-17 10:43 Very good tutorial, thanks Comment by Vadim 8212 2011-10-11 19:42 man, this is the best introduction (covering all topics) into DSP I stumbled upon perhaps I do have a chance to pass the exam. D sorry, for a double post. but8230 can you attest everything is correct for example, 822082218221 You can use any number of poles and zeros, but they must all have 8220conjugate pairs8221, if they are not positioned on the 8220-8221 axis. 822082218221 is this true I8217m playing with applets that allow for poles without conjugate pairs and seemingly band-pass filters (with regard to the magnitude response) can be built this way. can you please explain ( laps. fri. uni-lj. sidpsarhivappletiisipsystemv4.0srcapplet. html ) Doug, it is true, IF you want the filter to have a real output, not complex. If you make a bandpass with just one pole, and have the pole so close to the unit circle that the filter output is pretty much a single frequency, then the output of the filter will be a complex phasor rotating in one direction on the complex plane. If you switch the sign of the imaginary part of the position of the pole then you get as output a phasor that rotates in the opposite direction. If you have poles in both of those positions, then the output must contain both of those complex phasors in equal parts, thus the imaginary parts of the phasors cancel each other. So you get as output a real sinusoid. Good luck with the exam Comment by Olli Niemitalo 8212 2011-12-27 13:41 This is the first cogent explanation of poles and zeros that I have ever received. I feel better and worse at the same time, if you know what I mean. In any case. THANK YOU Comment by Mark McConnell 8212 2012-05-09 01:12 8230 Yehar8217s Digital Sound Processing Tutorial for the Braindead 8230 Nice Job Men82308230. I found it very helpful. Terima kasih. Can you put implementation of audio effects in computer. Comment by Trnform3r 8212 2012-09-16 10:07 Sure, for example as a VST effect. Comment by Olli Niemitalo 8212 2012-09-16 22:14 This is fantastic nice work and a very well explanation of DSP. Thank you :D Comment by tor 8212 2013-02-16 01:42 Thank you so much for this informative writing on the subject which makes life much easier since no-where could I find any book on the subject which makes it as clear as you did here. Keep it going and thank you again. Comment by FJ Botha 8212 2015-02-21 10:14 Frickin delicious Seriously, i thank people like you for simply existing and count my blessings that i found this brilliant introduction you created. The note takingoutline is digestable in one bite and it will stick with me during my upcoming solo winter sound holiday to the pampa and magellians strait, the large uninhabited Falkland rock, and if im still alive - christmas island. Dec to Feb. I hope to capture enough sound to keep me glazed and deadeyed until black metal villians capture Oslo Comment by Mick Dkaye 8212 2016-10-18 19:13 And love that Black Deck. Masonna weeps Comment by Mick Dkaye 8212 2016-10-18 19:16 Leave a commentDSP filter - DISPFIL. EXE Version 1.09J - June 18, 2000 JE3HHT Makoto Mori Translated into English by JA7UDE Oba This is a DSP filter tool using a PC with the soundcard. With this tool, you even can design various types of digital filters including adaptive filters. However, this tool is just experimental and will not afford the practical use for amateur ham radio. You probably need a powerful CPU to make this tool run flawlessly. In addition, you need a soundcard with the full-duplex mode. I made this program just because of my own interest. As I was not quite familiar with the use of the soundcard, it still has substantial time lag from input to output and might not well work for CW. Needless to say, this program is freeware. How to uninstall DSPFIL does nothing to the Windows registry, so just delete all the files with the directory that has DSPFIL files. OS: Windows 95, 98, NT (Note - by VE5KC - has worked fine in tests with Windows XP amp Vista) PC: The faster the better Display: 640 x 480 or more Soundcard: 16-bit soundcard that is capable of FULL-DUPLEX (some cards wont work) Hookup and Operation Connect the speaker out of the radio to the Line-in or Mic of the soundcard. Connect a headphone or speaker to the output of the soundcard. Since the Mic input has too high gain, I recommend the Line-in. Adjust the input level by using Mic or Line level in the Record property or the audio in the control panel. You can do that by using the AF gain of your radio, too. Adjust the output level by using Wave or Master level in the Play property or the audio in the control panel. You also can do that by using Up (up arrow) or Down (down arrow) button on the DSPFIL window. If you have a sound output from your speaker without running DSPFIL. EXE, your PC is configured so as to play the recording signal directly and thus you must turn it off. Go Play property and get Mic-in or Line-in muted. If you hear a sound immediately after starting DSPFIL. EXE, you are ready to go. In case you see a message like quotCannot open the sound device, quot your soundcard probably does not support the full-duplex mode. Give up listening to the filtered sound, but you can observe how DSPFIL. EXE works by the FFT and adaptive filter response windows. Since there is a time lag between the input and output, you should keep the buffer size as small as possible. The time lag has a big trouble in filtering CW signals (you will soon understand what it is when you transmit a signal, Hi). Too high input level causes distortion in the analog circuit of the soundcard. You have to adjust the input level by monitoring the FFT display set to quotIN. quot When overdriven, DSPFIL shows quotOverquot in the upper right corner of the FFT window. When the HPF button is depressed, the 100Hz high-pass filter is activated to the input circuit. It is effective if you have DC ingredient, but it raises the CPU load. Use it only when you need it. Details of the filters This is a comb filter using moving average. This filter, by its structure, gets the actual center frequency Rfo shifted from the defined center frequency Fo by RFo fss int(fssFo) Hz ifss fs 2j This can be compensated by carefully choosing the sampling frequency (fs). However, the sound blaster card does not allow fine tuning around 11025Hz, so DSPFIL admits the shift, Hi. The filter does not use a simple averaging calculation but uses subtractions for 12 periods. Thus, the even harmonics are suppressed, but the odd harmonics can be passed through. It is a good idea to use a 500Hz filter of your radio. It has lower quality in the frequency domain compared with BP100, but I think this filter gives the best performance particularly for weak signals. This is a band-pass filter using an FIR filter. It uses x3 oversampling. The physical sampling frequency is 11025Hz while the application sampling frequency is 3675Hz. If the number of taps is increased, the filter become sharper. However, it increases the processing time at the same time, and therefore it will not run on a slow PC. This is an adaptive band-pass filter for CW. I have not tested a lot on the values of fnofEcirc(mu) and fnofAacute (gamma), but I think the filter works, hi. This filter does not affect Fo or Tap, which is configured in the main window. The frequency-domain graph in the lower right corner shows the frequency characteristics of the transversal filter calculated with the coefficients, which are changed by LMS. You can see how the adaptation is performed by changing the frequency of the input signal. In case of weak signals, the filter coefficients tend to be small, which would result in a low level output. To compensate this, LMSB2 leaves the AGC turned on to increase the volume for the weak signals. This is a fixed frequency BPF for SSB. The low-cut frequency is fixed to 200Hz. If it oversamples the 2.2KHz or higher signals, it causes folding errors because of the decimeter. This filter does not affect the Fo, which is configured in the main window. This is a noise smoother for SSB. The adaptive operation might not be well tuned yet. The SSB signal is smaller autocorrelation than the CW signals, so I put small values in the correlation delays. This filter does not affect the Fo or Tap, which is configured in the main window. This is an automatic notch filter for SSB. It would give better results if it had faster convergence behaviour. However, I dare to focus on the response speed for CW signals. This filter does not affect the Fo or Tap, which is configured in the main window. This is a user-customizable filter. The default setting gives a wide band-pass filter for SSB. You can customize it by pushing the DESIGN button (the button face text is written in Japanese). You can copy the parameters of the selected other filter to those of this filter by pushing COPY (the button face text is written in Japanese) button. This filter does not affect the Fo or Tap, which is configured in the main window. User setting for the adaptive filters LMSBP, LMSNS, LMSAN are built-in filters, but the user can design a LMS filter by himself. Push DESIGN (this text is written in Japanese, so it might not correctly appear with non-Japanese Windows) button and select LMS, then push UPDATE (in Japanese text) button. Now you can change the parameters. The algorithm used in the adaptive filters is called Leaky LMS (Least Mean Square method). The user customizable parameters are: Tap the number of orders of the transversal filter Delay the number of delay nodes 2u the response speed V (gamma) the dumping factor Larger u gives faster response but slower convergence. Smaller V (gamma) makes the coefficients decrease faster when the input signal is cut off. However, too small V (gamma) will result in oscillation. Generally put a value a little bit smaller than 1 to V (gamma). If the REVERSE OUTPUT (in Japanese) is checked, DSPFIL outputs an error signal. It is checked to design an automatic notch filter. When AGC is checked, DSPFIL automatically increase the output volume for weak input signals. The characteristics of the adaptive filters are dependent not only on u (mu) and V (gamma) but also on Delay and Tap. Change all of them to see what happens. Use parameters that are given by another design software If you want to test the filter coefficients that are calculated with another filter design software, try the following steps. 1. Push DESIGN button in User1. User6 2. Select User and push Update 3. Push SAVE button and make a filter definition file. 4. Exit DSPFIL. 5. Edit the filter definition file using a text editor. Sampling frequency No over sampling 11025Hz OverSmp1 Over sampling x2 5512.5Hz OverSmp2 Over sampling x3 3675Hz OverSmp3 Filter order Put the number of orders to TAP field Coefficients Put coefficients to H0. Hn (n Tap) fields 6. Start DSPFIL and push one of User1. User6 7. Push LOAD button to load the definition file. CPU Power Since this program intensively uses floating-point operations and is not well optimized, it will not run on a PC with a slow CPU. 73 de JE3HHT Makoto MoriLoad error in equalizer data in some of the 8.21 builds, due to the 62986299 equalizer frequency fixpressors: Added Feedback2 mode. Different behavior from normal Feedback mode, Ratio is . New product: Thimeo WatchCat. Avant version added. RDS encoder: Added code 0 for FM 87.5 in the AF list. Seems to be supported by some receivers. Stereo Tool: Added info to. sts files to allow importing them in Omnia SST. SST: Added possibility to import Stereo Tool settings. SST: Attack and release are now faster when slider moves to the right. SST: Attack Multiplier: Attack is now faster when slider moves to the right. SST: Improved naming of many things, to be compatible with other Omnia products. SST: Disabled MULTICORE for latency 128 and 256, and LQLL. SST: GUI: Added glowing yellow line around SST tray icon, to make it more visible. Bug fix: ACR Stereo caused some mild crackling. Bug fix: RDS quadrature mode could cause crackles and overshoots. Bug fix: SST: Loading a preset stopped server communication. Some changes could get lost due to this. Bug fix: MAC VST version didnt work in OcenAudio (actually a bug in OcenAudio). Bug fix: SST: Multiband text below horizontal meters was upside down. Bug fix: Restarting on unhandled exceptions didnt work in 64 bit version. Bug fix: VLC: potential crash when switching between streams due to memory allocation issue solved. Bug fix: 1 channel audio processing was broken in 8.20. Bug fix: More-than-2 channel audio processing was broken in 8.20 due to what appears to be a compiler bug. Bug fix (Stereo Tool, SST, Omnia.7. 9. 9sg, possibly .11, BreakawayOne): Time zone changes ignored after startup on Windows (compiler runtime bug). Bug fix: BS412: Singleband compressor was placed after BS412, should be before. Bug fix: SST: Reset settings when loading preset, it kept the current settings if they were not present in the file. Bug fix: UECP client could hang after connection close. Now a new incoming connection kills the existing one. Bug fix: Equalizer frequency could be either 6298 or 6299 due to rounding errors, causing preset load errors. MicroMPX: New license mechanism for ARM version implemented. Code improvement: Less duplicate code in binary for Compress. Smaller binary, faster build, better performance. Added option to run Stereo Tool as a service. RDS: Moved reading RDS from file to a separate thread, should avoid hiccups when reading from network drives. Added code to handle (restart) and log crashes in sound card drivers and VLC. Not yet fully tested. Linux: Now using Ubuntu 14.04 instead of 16.04 to build (better backward compatibility). Linux: Now statically linking libstd (better backward compatibility). Linux: Switched to newer compiler for better performance (20, on both Intel and ARM). Web server: Whitelist allows as in 192.168.. instead of 16 (Leifs formatting), always allow 127.0.0.1. Compressors: Volume jumps on preset changes reduced (compressor adjusts to changing target level). Declipper: Detection (veil) didnt always work properly with gt 60 kHz input sample rate. Performance: Reduced CPU load of IO and processing framework. GUI: Hovering over tray icon now shows version number. GUI: Changed scale of many sliders, dB scale sliders are now flat over the whole range. Bug fix: Potential sound card glitches (bug introduced in 8.12). Bug fix: Crash reported by Camclone Bug fix: FM test tones were broken when using Ignore high frequencies (bug introduced in 8.12). Bug fix: VST plugin no longer cuts off the start of songs (bug introduced in 8.14). Bug fix: UECP port was always used (even if FM disabled). Bug fix: Potential bug in non-ASIO code could cause crashes depending on compiler optimizations (never seen). Bug fix: Potential crackle in lowest multiband band fixed. Rewrote error logging code. Omnia 9sg: Builds with the new framework now. SST: Make IPPORT changable without web gui. SST: Preset list was empty on first start SST: Improved naming of settings in Processing section. SST: Web interface code compiles separate from SST now. SST: Added importing Stereo Tool presets. SST: Multiple MicroMPX output IP addresses and ports CN. SST: Launch browser automatically on first start. SST: Improved startup screen. SST: Currently active presets highlighted. SST: Prepared for longer list of MicroMPX IPports (partially there). SST: Clicking on meters in top bar takes you to the page directly. SST: Top bar meter that corresponds with active screen is highlighted. SST: Add glowing yellow line around SST tray icon, to make it more visible. MicroMPX decoder: Buffer underruns seen on ARM version, caused by loggingtracing library. MicroMPX decoder: Shows more startup info and waits for pressing ENTER at end (command line). MicroMPX decoder: Improved handling of out-of-order and lost packets. Added built-in self test for sound cards. SST: Slider behavior improved (slow vs fast moving the mouse). SST: Preset list was sometimes empty, message added that in this view no presets are available yet. SST: Preset list jumped to new view after opening which looked weird. Fixed. SST: Composite Clipper is on by default. SST: Added presets. Bug fix: SST: Rules are made to be broken disabled when not using dual side band. Bug fix: SST: Clip LR peaks above was always disabled. Bug fix: Glitches due to missing thread priorities fixed (since 8.12). Bug fix: Non-2 channel audio (1, 5.1, 7.1 etc) didnt work (since 8.12). Bug fix: Stereo Tool web interface was adding 20 marks in text boxes when typing spaces (since 8.12). Bug fix: ASIO in 192 kHz, non-ASIO Normal output set to input audio got 48 kHz audio causing underruns (8.12). Bug fix: CPU core affinities Main 1 and 2 were overwritten by LQLL affinity settings (since 8.12). Bug fix: Volume drop when using old Calibration settings (since 8.12). Bug fix: Heap corruption on close for command line version (only visible when a debugger was attached). Bug fix: Linux: Web interface exceptions when closing the web interface. Bug fix: Weird crash in Adobe Audition 3.0 (from 2007), gone after changing compiler settings. Possibly bug in Audition or compiler. Bug fix: Separate thread for SCA input with ASIO didnt work, disabled that option now. Bug fix: GUI: ASIO inputs sometimes showed dotted overlay moving from half-way to top (and disappearing). Bug fix: GUI: Values for Max Attack and Max Release were displayed incorrectly. Bug fix: ASIO buffer was too small, could cause glitches after a buffer underrun. Bug fix: Improved sound card synchronization constant clock speed determination. Bug fix: Sound card timeout times increased to take VLC connection time into account. Bug fix: VST version didnt work in Wavosaur. Cause unknown but it works now. GUI: Gate vs Gate Freeze ranges were different, causing sliders to go outside of their range. VST plugin: Process function crashed when called with lt 0 samples (Wavosaur empty file bug). Added, removed and updates presets. Framework: Added support for upto 7(7) threads, works much better on slower CPUs with many cores. Framework: Added Quality settings 100 (OmniaSST only). With this, latency can go down to 128 samples (5 ms analog input to analog output). Framework: Smooth preset switching - no more hiccups if the same filters are used, also faster. Linux version: High CPU load (100) in one thread fixed. Linux version: Made plugin for Rivendell (and other programs that want to use it). Not yet fully tested. Linux version: Didnt run on Fedora. Fixed. Memory usage reduced. Sound cards: Resampler improved, now measures separate buffer filling and speed offsets. AGC: Added Gain slider. AGC: Added Windowing. Compressors: Added Windowing. Composite Clipper: Rules are there to be broken mode could cause occasional 1-sample overshoots. Fixed. Low Latency monitoring: Affinity setting added. Low Latency monitoring: Latency selection added. Removed lower lowpass filter for high frequency input at lower latencies, not needed anymore. GUI: Showing AGC gating the same way as for compressors GUI: Low Latency monitoring: CPU usage display added. GUI: Added Hear buttons to listen to the output upto this point. GUI: Fixed synchronization issue between meters, scopes and audio. GUI: Help pop-ups also work on Mac OS X now. GUI: Renamed PNR Noise Hum to Dehummer. Web server: Multi-threading added. Now handles upto 10 requests simultaneously. Web server: no-cache added to objects. js. Web server: Automatically add index. html if a directory is specified. Web interface: Added white list for logging in. Web interface: Added spectrum display for Delossifier and Absolute Highs. Web interface: Buttons didnt work. Generic plugin: Added option to make the plugin access input and output sound cards (Windows only). Bug fix: Potential crash in ASIO when changing channels fixed. Bug fix: Potential crash when zooming waveforms fixed. Bug fix: Latency with ASIO was not fully predictable on restarts. Fixed, allows for slightly lower latencies. Added Delossifier, repairs MPEG-compressed audio. Added Absolute Highs, generates lost (lowpass filtered) high frequencies. Added FM MPX Analyzer with MPX demodulator, RF bandwidth view and LeftRightQuadrature () decoder. Added RDS ASCII external input support for programs such as Arctic Palm, RDS Magic. AGC: Added Ratio control. Composite Clipper: Added Rules are there to be broken mode for extra loudness and clarity. Clipper: Added Multipath Stereo Phase Difference Remover to improves reception and reduce multipath. FM: Added Polar stereo encoder for OIRT 65-74 MHz (Soviet system) FM stations. FM: BS412: Filter bass based on ITU.1770 instead of the existing kinda-randomly-chosen weighing. AM: Added test tones, including a sawtooth with a pauze at 0. Improved resampling between sound cards (more improvements are still needed). Raspberry Pi: Optimized the code a bit. Linux versions: Remove wxWidgets, replaced by pure XLib. Mac OS X versions: Remove wxWidgets, replaced by pure Cocoa (VST: only 64 bit so far). GUI: Auto-hiding deprecated pages and sections if they are disabled: AGC: Compatibility amp behavior header Bass AGC (Old) Classic Multiband Compressor Classic Singlenband Pre-limiter (no longer needed since Leifs clipper improvements) Advanced Clipper: Bass protection (Deprecated) heading Advanced Clipper: Highs gap protection heading (replaced by Leifs gap protection) GUI: Auto-hiding deprecated settings if they are set to the optimal value: Leifs clipper efficiency modification checkbox (overruled by Leifs clipping strictness) Composite clipping Strictness (CPU) checkbox Phase rotation: At start of processing checkbox All Singleband and Multiband Compressor type (Analogdigital) pull-downs Multiband and Natural Dynamics Flat Frequency Response Compatibility mode (bad) checkbox Advanced Clipper: Smooth slope setting (under Distortion) Advanced Clipper: Upsampled highs clipping (useless) setting (under Distortion) Advanced Clipper: Stokkemask: Force Stokkemask even if not using Composi te Clipper (bad for audio) BS412: De-basser non-ITU.1770 compatibility mode (bad) checkbox GUI: Added tool tips. GUI: Add indicator that a setting has been modified. GUI: Added reset mode to last-loaded setting beside the existing reset to default. GUI: Made scope skinnable. GUI: BS412 disabled Advanced Clipper Drive. GUI: AM: Added Forcibly remove DC offset to the AM settings panel. GUI: Hide Stereo Tool and Thimeo names in whitelabel builds. GUI: Made distinction between switches and onoff buttons. GUI: Added Multiband Quick Overview screen with the most important settings. GUI: Redesigned Simple mode pages: Single screen overview. GUI: Added names Diversity Delay and HD to indicate that it can be used for HD. GUI: Made R3LAY skinned version for LAWO R3LAY. Command line version now also handles licensing properly. Bug fix: Fixed VST DLL unload bug that affected Adobe Audition. Bug fix: Fixed crackle at startup caused by Ignore High Frequencies (even if disabled). Bug fix: BS412 was still active when FM was disabled. Bug fix: Sound card setting Input without processing didnt work in stand alone version. Bug fix: ASIO: Fixed 1624 bit LSB modes used by new AudioScience 192 kHz card. Bug fix: ASIO: Added a check to get rid of a small memory leak in the ASIO SDK. Web interface: Added Access-Control-Allow-Origin: to binary HTTP protocol to support cross-server scripting. Generic plugin API: Added GetApiVersion function. Generic plugin API: Added stereoToolIsLicenseValid function. Added, updated, removed and reordered presets. Improved quality of bass, ringing artifacts cleanup filter caused some clicking effects in low bass (Since v 5.00). Added 8x oversampled non-phase linear Butterworth de-emphasis to match some hardware units pre-emphasis. Sound cards: Added backup sound card support (sound card 1 switches to 2 after some silence). True Bass: Add highpass filter for lower latencies because it can cause very low subs, caussing problems on FM. Highpass filter: Improved non-phase linear HPF a bit, works better when used after other filters now. ASIO: Only claim the channels that are actually in use. ASIO: Add already in use as possible cause to ASIO claimBuffers error message. ASIO: Added LeftRight indications in dropdown list. ASIO: Changed default channels to disabled except for Normal Output and Input. ASIO: Add more output selections - Increased from 8 to 32. Ignore High Freqs: Added Auto frequency option. Determines lowest possible value for max CPU load reduction. PNR: Settings now stored in. sts file for VST plugins (partial fix of the problem). Clipper: De-esser: Changed default value, now disabled by default. Reduces the CPU load. GUI: MPX display disabled when waveform display is disabled. GUI: Improved RESET. now gives preferred values instead of traditonal values needed for old preset compatibility. GUI: Disable Composite Limiter slider when Hard Limit is disabled. GUI: Bug fix: Phase Delay was not grayed out in Bypass mode. Bug fix: Some settings were net reset when loading a default preset in previous version. Bug fix: VST: GUI-related hiccups in Adobe Audition. Bug fix: Synchronization to external RDS signal could sometimes have a hiccup due to a threading issue. Bug fix: BS412 level calculation was wrong for LR instead of MPX output when using non-Pre-emph output. Generic Plugin: Add function to Get parameter values. Added new presets. FM: Add extra pre-emphasis for clipping for stations that send the processed FM signal through a lossy codec. Web interface: Added Presets dropdown menu and Bypass in web interface top bar. Sound card control: Slightly better low latency handling for non-ASIO. Sound card control: Auto synchronization mode means OFF when any of the affected buffer sizes is lt 250 ms. Bug fix: Error in AM AF frequency list for RDS. Some high AM frequencies might not have been encoded correctly. Bug fix: Simple Clipper Pre-limiter amplification was performed even when Simple Clipper was disabled. GUI: Bug fix: True Bass was called WARM BASS in the LOADSAVE menu, and WB in bottom menu. GUI: Bug fix: Lowpass filter frequency was displayed incorrectly for Separately processed streaming output. GUI: Bug fix: Pre-emphasis for MP3 was shown disabled when using Separately processed streaming output. Code: Redesigned for faster compiling. Updated presets. Generic plugin: Added function to set parameter values. Generic plugin: Added export of ParameterEnum for generic plugin customers. Bug fix: In 7.80 Automatic restart could be triggered incorrectly when the buffer size was very small. Bug fix: Fixed some small issues caused by the Thimeo framework rewrite, had no effect. FM: Improved Stokkemask (ITU-R. SM1268) filter for composite clipper (sounds better). FM: Fixed several bugs, mainly noticeable in SSB mode, at lower Quality settings. FM: SCA1 channel is now displayed in MPX graph. Updated and added some presets. FM: Added composite clipper FM: Added SSB (Single Side Band) mode. (Requires composite clipper.) FM: Added Multipath clipper. (Experimental Requires composite clipper.) Loudness: Improved Airy Highs. Highs now sound crisper and less tonal. Loudness: Rewrote clipper to reduce CPU load (Strictness can be set lower to give similar results). Added Phase equalizer. Updated and added many presets. User interface: 5 modes ranging from Simple to Extreme Tweaker. Basic should be good for most people. Greatly reduced CPU load of almost all filters. With identical settings, the CPU load is typically 12 reduced. Singleband compressor completely redesigned. Keywords: Attack, release, release hold, ratio, feed forward vs feedback, lookahead, burst protection. Multiband compressor completely redesigned: Configurable number of bands monitoring individual bands, median display. Added graphics equalizer. AGC: Added side chain. AGC: Added singleband compressor after AGC. AGC: Smoothed Band 1 to band 2 coupling. AGC should work better if declipper or noise gate is activated. Extra lowpass filter before processing to fix problems with songs with excessive very high frequency content. Set 0.25 kHz above actual LPF freq, but never below 16 kHz. Loudness: Improved delayed bass clipping protection to cause less distortion. Provision to discard buffer after a certain period of inactivity added for VST plugin. FM: SCA was missing since version 7.00, added again. Declipper: Reduced artifacts when using declipper in combination with non-phase linear highpass filter. Declipper: Changed default settings based on input from Jesse Graffam. Sound cards: Synchronize to output now also works with Normal output. User interface: Made 6 dB lines in meters more prominent. Bug fix: Disabled mouse hold behavior for all widgets except sliders. Updated and added many presets. Loudness: Added delayed bass clipping for better punch. Loudness: Fixed Bass Shape artifacts, and now also works on higher frequencies. Loudness: Added option to dynamically switch off distorted midshighs for tones. RDS: Better special character display in GUI. Updated presets. Added 5.1 and 7.1 audio support (channels are declipped, highpass and lowpass filtered and clipped, nothing else yet). Loudness: Improved bass clipper, better punch and warmer sound with more mid-bass. (old vs new) Loudness: Removed volume drop between 2.8 and 3 kHz. RDS: Added convertor from Windows-1252 to RDS characters to support special characters. Installer made smaller by moving non-SSE2 version to a separate installer. Now below 1 MB again :) Bug fix: Drawing issue in 7.00 and 7.01 pull-down menus without scroll bar when using mouse wheel. Bug fix: Left and right channels were sometimes swapped when loading a preset in 7.01. Bug fix: Stokkemask clipper was broken in 7.01. Carbon Coder plugin now supports R128 normalization for 5.1 and 7.1 audio. Added some presets. Solved several VST GUI issues in several VST hosts. Made VST window a bit bigger. On loading built-in presets, most FM and sound card settings are no longer reset. Volume increase on preset reload should not occur anymore, although Im still getting reports that it does. But at least one cause is solved. Added sound card restart button, was missing in version 7.00. Bug fix: GUI unresponsive at very high CPU loads when you click right of text in text box. Bug fix: BS412 acted weird with FM sound card enabled, pre-emphasize output disabled at low input sample rates. Updated presets. Total GUI redesign. Interface dynamically adapts to screen size and is nearly touch screen ready (keyboard needed). Added new version alerts and specific warnings which unregistered functions are in use. Last selected preset name visible after restart. Added web interface (no access protections yet). Completely rewritten parameter handling code solved bugs and reduced the chance of introducing new ones. Added ability to reset, load and save parts of presets. Added input sound card Gain control. FM transmitter audio synchronization added to synchronize streaming audio at multiple transmnitter sites (experimental). Loudness: Changed bass clipper behavior, flatter frequency response sounds a bit warmer. Loudness: Bass clipping filter bug fixes: Whistle artifacts gone, knock-on-wood artifacts reduced. Loudness: Airy Highs turned off if there are very little highs (reduces crackling sound that sometimes occurs). Declipper: Reduced clicking artifact in reconstructed audio. Declipper: Improved splitting load over 2 cores. FM test tones: Added Bessel null frequencies and smooth square wave (sounds less loud) test tone mode. Removed a number of memory allocationsfrees that occurred during processing. Added a lot of presets. Natural Dynamics is available in the GUI again, if you enable it explicitly in the INI file. Added sound card input and output tilt adjustment (2nd order). Loudness: Changed clipper distortion masking model, giving less distortion at nearly the same highs level. Loudness: Added Airy highs, makes highs sound both louder and more natural (less restrained). Loudness: Removed bugs that caused hole punching by loud highs. Loudness: Highs Priority no longer causes volume drops caused by loud highs. Loudness: Redesigned de-esser for pre-emphasized sound. Bass sensitivity: If there are both mids and highs, the drop is a bit less strong than before. Bass sensitivity: Determining more precisely when the clip is needed. BS412: Added max deviation (- 75 kHz) display. BS412: Added headroom slider. Determines margin for error to avoid sudden volume drops. BS412: Max deviation can be lowered to create denser sound and protect against sound card problems. BS412: Faster rise after fast drop for louder end result. GUI: Added displays so you can see what is being sent out to the sound card (including calibrations). GUI: Cleaned up Loudness panel (removed some things that are no longer needed). Bug fix: Latency 512 didnt work properly. Bug fix: Fixed GUI issue where singleband compressor settings were not always correctly loaded. Bug fix: AXIA input sound card loading issue. Added Carbon Coder plugin with multi-pass R128 support. AGC: Added a slider to protect against volume drops for loud voices. Loudness: Improved overall filtering (removed post ringing) at latency 4096, sound is cleaner in many ways (especially bass). Added some new controls for configuring it (Time Spread) Loudness: Advanced Bass Distortion Protection: Improved bass clipper. Bass is now louder and clipped tighter Loudness: Added bass shaping. Adds harmonics to very low frequencies to make them sound louder Loudness: Allow turning highs intermodulation distortion off Loudness: Added highs priority slider, increasing this reduces high frequency intermodulation distortion but may cause some drops in other frequencies Loudness: Asymmetric bass reduction should not go below dynamic threshold Loudness: Reduced strictness, the high precision wasnt really useful Loudness: Split sensitivity for bass clipper: Can now be configured separately for mids and highs. Result: more constant and higher bass level Loudness: Turned off old bass protection mechanisms in default settings. Caused artifacts, no longer needed Loudness: Reduced clipping strictness for mid-bass frequencies (lower CPU load) Loudness: Increased de-esser range, and it can be turned off now Loudness: Simplified clipping algorithm Stereo Boost disabled for lowest frequencies (below approx. 140 Hz) BS412 limiter: Show a warning and info when BS412 is enabled BS412 limiter: Fixed drops issue. Very brief drops can still occur but they are usually hardly noticeable BS412 limiter: Changed default values for more constant sound level Added and updated many presets Split values for Singleband, Singleband AGC style and BS412 Bug fix: Stereo Image was broken for latency 512 Bug fix: Icon now visible in stand alone version Bug fix: Stokkemask clipper didnt work at latency 512.Major code cleanup and restructuring Added BS412 limiter Added Stokkemask clipper Improved scope display (triggered) Improved low latency (6 ms) monitoring output in Stand Alone version Improved overall quality by changing the processing window shape for ALL filters Added support for cheaper limited WinampDSP declipper license Reduced CPU load when playing pure silence AGC: Improved handling of sudden loud bursts Loudness: Improved protection of mids against loud highs, which were causing drops (holes) in the sound. Loudness: Changed the bass protection shape Loudness: Added protection against distortion caused by asymmetric bass sounds Loudness: Slightly better protection of frequencies below 700 Hz against very loud bass Loudness: Bass protection and de-esser levels were calculated incorrectly if Final Clipper was enabled with a volume other than 1.00 BS412 limiter now also works with Pre-emphasize output disabled. Phase rotator: Improved behavior and display Bass AGC: Now works on a much smaller frequency range so it only affects really deep bass sounds Bug fix: Changed default values for the Allow more distortion sliders in the VST version Bug fix for Omnia 9: RDS AB flag works properly now, and if you change the RadioText and there was no sequence of texts there the new text will start to be broadcast immediately Bug fix: Forcibly remove DC offset in Loudness did not always work properly Slightly reduced Loudness CPU load, but the new highs protection adds a lot more if its enabledAdded multicore support for Declipper and Loudness. Throughput is now 38 higher in presets that use both. Loudness: Fixed a bug in the non-SSE2 version. Declipper: Fixed memory bug in clipping detection. This may have lead to weird issues. Improved performance when closingminimizing Stereo Tool if the virus scanner NOD32 is used. Bug fix: Loudness de-esser setting was not saved in VST plugin. Bug fix: Loudness values above 3.00 were not loaded correctly in VST plugin. Bug fix: TA button in main window did not work. Changed a value in Bojchas Neodymium preset. There is now also a separate declipper program See perfectdeclipper. Added declipper filter Loudness: Completely redesigned for more consistency, better transients, louder output and greatly improved highs. Punch is no longer needed. Loudness: Improved vibrating voices filter caused by loud bass. Loudness: New filter that removes highs distortion caused by loud bass, now allows bass levels upto 100 without (much) distortion in the highs. Loudness: New filter that detects when highs cause volume drops in the rest of the sound, and briefly lowers the highs level. Loudness: Added oversampling checkbox for streaming and mastering: This makes sure that no clipping will occur if the audio is upsampled afterwards. Loudness: Added separate dirty bass mids highs sliders for better bass control and artistic effects. AGC: Redesigned to handle different input levels much better. AGC: Enabled ITU-1770 support (for bass and head separately). AGC: Added stereo separation control. AGC: Added start level for mastering. AGC: Separate Remove remaining peaks settings for lows and mid-highs. AGC: Added Force level below slider for improved peak control. Forces the AGC level down faster if the difference gets too big. Added Bass AGC filter to reduce excessive bass sounds. Multiband: Added voice protection settings to better cope with loud vocals in music. Multiband: Allow changing band frequencies (only via STS file). Multiband: Added control of link between bands 1 and 2. Pre-emphasis: Moved to end of processing, just before clipping. Highpass filter: Improved filtering to reach a cleaner output. Highpass filter: Added order setting for non-phase linear filter. Improved audio processing window shape to reduce artifact levels. Improved pre-ringing filtering to avoid artifacts that occurred before. Added SCA output channels (SCA1 works, SCA2 is problematic due to the required frequency range). Support added for multiple Stereo Tool installations with different settings per installed version (requested by Jazler). Added registration to command line version. Bug fix: Fixed Test right channel in FM Transmitter Calibration (broken in version 6.10). Bug fix: FM Calibration frequencies 23-57 kHz were not at the correct frequencies. Bug fix: Fixed leftright channel swap that could occur in the stand alone version. Bug fix: AGC could hang in some cases when enabling more bands. Fixed. Bug fix: Phase rotation did not sound good at low latencies in version 6.10. Bug fix: Time zone in RDS was wrong. GUI improvement: Made waveform background black. RDS: Adjusted behavior if no special characters (lt, gt, , ) are used: Word wrap with centering is selected by default. This was done for the Omnia 9 that uses the Stereo Tool RDS encoder the behavior is now identical for both systems. RDS: Ignoring trailing spaces for word wrap. Added a lot of presets. Added interface for blind users. Set environment variable STBLIND to 1 or YES. Reduced latency: Latency: ASIO: Add processing priority control to enable very low sound card latency ( 1-3 ms) without hiccups total latency of 16 ms possible. Latency: Upsampling (for FM) and downsampling (for high input sampling frequencies) no longer causes extra latency. Latency: Reduced Composite Limiter latency from 1.7 ms to 0.9 ms at latencies 512 and 1024. Some (very small) effect on audio. Latency: Found a bug that caused hiccups in audio processing, fixed it. Results in lower possible latency. Latency: Pushing data to the output before the processing instead of afterwards to reduce latency. Improved audio quality at low latencies: Dynamically adjusting audio processing windows at low latencies to better match the type of audio that comes in. Dynamically adjusting clipping effectiveness to avoid clipping too much in certain conditions, depending on audio processing window and audio content. This protects soft highs against horrible vibration caused by bass sounds at latency 512 and (less) 1024. Upsampling and downsampling: Far less artifacts caused, far cleaner output signal. Created alternative Phase Rotation filter for low latencies (512, 1024). Multiband: Added an artifact protection step that limits the maximum difference in reduction between adjacent bands, if they cause too much (configurable) artifacts. Multiband: Adapted frequency bands and band content to make Multiband at latency 512 sound as much as possible as higher latencies. Loudness: Disabled Improved Bass Distortion Protection filter for latency 512 because it caused distortion. Loudness: Fixed vibrations caused by Very deep bass distortion protection at latency 512 and 1024. Now also enabled for latency 512. Hard Limit: Fixed (minor) artifacts in upsampled audio. Bass Boost: Reduced artifacts at latency 512. Note: Using a less steep filter (bigger difference between frequencies) helps Other changes: Loudness: Improved Punch filter: Now causes far less artifacts than before. But it is less effective for soft sounds. Loudness: Removed distortion caused by Punch. Bass Boost: Removed distortion. Quality and Latency are now part of presets, and STS files. Bug fix: Loading and saving Noise levels in VST version resulted in incorrect behavior. Bug fix: Displayed Multiband output levels were unreliable at low latencies, especially clipping. Bug fix: For high input sample rates, now removing very high frequencies that cause issues with Hard Limit. ASIO: Added ASIO Configure button, which allows setting ASIO parameters such as ASIO buffer granularity. ASIO: Changed buffer size configuration to match whole ASIO grains. Added and updated a number of presets. Noise gate: Fixed crackling sounds in first band. Added FM Hiss filter. Removes FM hiss from input signal, useful for re-transmitting FM signals. AGC: Added Based on volume before Pre Amp checkbox to make it easier to create presets with consistent behavior at different Pre Amp settings. AGC bug fix: Behavior was different at different input sample rates. AGC: Lock band 1 to band 2 if the bass is loud but not extremely loud it is not lowered. This avoids sound changes caused by the AGC, and results in a warmer, more natural sound. Multiband: Added Flat frequency response mode which works much better. Multiband: Added display of the median amplification, useful for configuring Multiband settings. Singleband: Added AGC style mode. Doesnt seem too useful though. Final Limiter: DIFF mode fixed. Final limiter is now automatically bypassed when Loudness is used (improves audio quality and reduces CPU usage). Loudness: Added filters to protect highs against loud bass sounds. The bass level can now be set much higher without causing distortion, even if Punch is used. Loudness: Added maximum mirror reflection to make the sound more natural (less metalic) and louder. Loudness: Fixed Improved loudness distortion protection artifacts. Loudness: Fixed Very deep bass distortion protection artifacts. Loudness: Removed Allow louder lows slider (no longer needed). Loudness: De-esser: Replaced Allow louder highs by an AGC. Loudness: Added a slider to increase the dynamics of the output (by allowing some more harmonic distortion). Loudness: Added a filter to protect certain sounds (female voices) against vibrating. Loudness: Fixed dynamic Loudness input level behavior of Punch Hard Limit: Added an extra slower Hard Limit phase volume comes up slower when the peaks are really loud. Added Lossy Compression (MP3AACOGG. ) output optimization filter for webradio stations. See this image for a display of the effect of different pre-emphasis values on 128 kbits MP3 files this image displays a frequency analysis (high frequency content means clipping), again 128 kbits MP3, Post Amp at 90. FM Output: Repaired the lowpass filter which sometimes caused artifacts. VST plugin notifies host of the delay. VST plugin: Bypass mode fixed. VST plugin: Removed separate window border, now loading inside host window. VST plugin now also works in mAirList. Improved handling of hiccups in the stand alone version when the CPU load gets very high. Presets: Made Generic and Web Radio presets sound much closer to the sound of modern CDs. Presets: Made FM presets sound much closer to that of popular hardware processors (more bass and highs, among others). Presets: Added new user presets, removed some obsolete ones. Command line version: Files larger than 2 GB can be processed now. Linux version added Both a standard command line version and one with a GUI. Cleaned up unused code: Reduced executable size by around 40, installer by 25. Reduced CPU load by 3-4. Reduced memory usage by over 4 MB (from over 24 MB to under 20 MB). Removed part of Core Duo support from the SSE2 version (no effect on performance, only increased file size). Removed SSE2 support from the SSE version (which was accidentally included in it). Fixed Windows Vista This program might not have installed correctly warning. VISIT SITE button was being redrawn continuously - fixed. SSE (Pentium 3) version: Replaced some generic code by SSE code (70 performance boost). Multiband: Very large differences between equalizer or soft limit settings in the lowest bands are handled correctly now. AGC: Improved handling of sudden spikes. AGC: Remove remaining peaks gives far less artifacts, sounds more stable. AGC: Introducted dynamic hold time to improve control of output level (much closer for tracks with very different dynamics contents). AGC: Fixed bug that could cause strange behavior durig absolute silence. AGC: Added possibility to work on 2 channels combined, instead of just operating on them separately. AGC: Added gating. AGC: Added 3rd band for improved highs control. Noise gate: Disabled 1st band because it caused artifacts. Completely redesigned the processing pipeline to reduce the maximum latency from over 1.5 seconds to at most 0.1 second, with a better sound quality. Made latency configurable in 4 steps: 512 (ca. 13 ms), 1024 (25 ms), 2048 (47 ms) or 4096 (93 ms, best quality) samples. Reduced CPU usage by more than 50. Replaced 3 quality modes (LO, MID, HI) by a continuous slider. Reduced memory usage by 17 MB. Replaced Pre Limiter by an RMS-based 1 or 2-band AGC, which controls the volume much better. Added a Bass Boost filter. Added Phase Rotation filter that greatly improves the sound quality of many difficult sounds (trumpets, female voices) when loud output levels are required. Added a filter that removes any pre-ringing (weird sounds before a sound burst) which was present in older Stereo Tool versions. Improved the Multiband clippers audio quality: Multiband clipping sound much cleaner now. Removed Trumpet sound filter from earlier versions that was intended to improve the quality of certain sounds, but also caused a lot of side effects (no longer needed due to the new phase rotation filter). Removed Multiband HQ mode (no longer needed). Made main window resizable and maximizable. Bug: No 2 Stereo Tool instances can be loaded in a VST engine based using the same Stereo Tool DLL file. A popup now displays this message and gives a workaround. Presets updated (now using Bass Boost and Phase Rotation where appropriate) or added. RDS without stereo no longer generates a 19 kHz pilot tone. In case of FM output, normal output is closer to the FM sound, and the FM sound is clipped better before FM Overdrive - using FM Overdrive now leads to a 3 louder output signal than before with better quality. Input and output volume meters blink in red to indicate clipping if the maximum volume is reached (input) or surpassed (output). Improved performance of waveform and MPX display. Improved dithering (final step in processing): If audio is not changed, dithering doesnt change the sound anymore. Bug fix: VST version now allows opening multiple instances of Stereo Tool (earlier versions crashed). Bug fix: Hangups when playing. WAV or lossless files in Winamp. Bug fix: RDS AF (Alternative Frequencies) was not turned off properly if no frequencies are specified. It might have worked, or no, depending on the receiver. Added warnings about sound quality when latency is reduced or FM pre-emphasis is enabled. Configure Audio and Display button blinks when audio is being removed due to the Clear buffer if no data arrives for setting. Waveform display indicates Bypass mode. Improved looks of tray icon. Bug fix: Bypass all in ASIO mode (stand alone) now works properly. RDS volume adjusted: It was slightly louder than the setting. Loudness: Added yet another filter (Very deep bass distortion protection) to reduce bass distortion and make the total sound cleaner. Loudness: Now bypassing Punch for frequencies below 45 Hz for cleaner output sound. Loudness: Improved handling of female voices and xylophone-on-top-of-low-strings sounds, leading also to an overall more natural sound. Re-organized Loudness window. Should be clearer now. Stereo Boost: Added a filter that removes excessive reverb that occurs in some songs. The result is much closer to the original sound, but still with strongly increased stereo. Bug fix: VST plugin version: Some version 4.00 parameters were not saved. Slightly improved some presets. Loudness: Added a filter to reduce bass distortion. Bass levels can now be set much higher without getting more distortion than before, or kept at the same level for better sound quality. Loudness: Added a deep bass boost filter that boost very low (up to about 70 Hz) bass sounds by about 1.5 dB (at 100 Hz the bass is about 0.8 dB softer though). Loudness: Added a limiter to the CleanPunch filter to remove distortion caused when setting it close to Punch. Moved CleanPunch default setting closer to Punch. Low Quality mode: Changed the processing that is used for low quality mode. Quality is much better than before, and the CPU load is reduced. Low Latency mode: Fixed some issues in the low latency processing. Latency is now approx. 100 ms lower than before, quality is much closer to that of normal latency processing. Noise gate: Noise reduction can now be set for each frequency band. Added a number of presets with more bass. Added Marinos Urban preset to Web Radio presets. Added new stereo widening filter: Stereo Boost. Updated most of the presets to use the new Stereo Boost filter. Reduced the minimum value for the gap detection time. Bug fix: No audio problem unless bypass is used solved. Bug fix: Too long delay in tray icon volume display solved. Improved Hard Compression: Dance presets. Bug fix: Under certain circumstances occasional hangups could occur. That should be solved now. Single core systems were not properly recognized and used code that was optimized for multicore systems. The CPU load on single core systems should be slightly lower now. Loudness filter: Improved the filter that detects high frequency distortion caused by very low frequencies. Low frequencies are now much louder, and distortion is greatly reduced. Loudness filter: Added a slider to choose between clean sound (with ringing artifacts) or punchy, as-close-as-possible to the original sound. Punchy sounds a lot better, but it may introduce some distortion (which is largely solved by the fix on the previous line). When opening a window that was already open, it jumps to the front now. Stereo Tool logo added to the window title bar, also displayed now when using ALT-TAB. Bug fix: Version 4.01 crashed on some Vista systems. Loudness filter: Improved bass sound, especially for high (3.0) Loudness values. Bug fix: VST plugins working again. Interface: When clicking on a panel which window is already open, it jumps to the front. Redesigned the interface. The main window is now smaller and less crowded. Each filter has its own window now, and many values are displayed in dB also. The output displays in the main window are now synchronized to each other, and can be synchronized to the audio player sound card output. Added a fix for the annoying part of the previous song gets played at the start of the next song problem in Winamp. Note that this fix is a workaround for what is essentially a problem of Winamp, and it may not be perfect. Reorganized the Presets pulldown menu, and added the 10 last used preset files to the list. Also the currently selected preset is visible in the Presets pulldown button now. Added more control over how the Loudness filter works. Its now possible to allow more highs or more lows to pass through (this may cause some distortion though). Pre Limiter is now using perceptive volume instead of peak level for much better initial volume control. Added a filter to remove remaining loud peaks after the Pre Limiter filter. The Pre Limiter works with a delay, the new filter is a lookahead filter that removes remaining peaks, which occur mostly at the start of a new track after some silence. Final limiter now has an option to use less CPU power (in that case it doesnt preserve the volume). Added a pre-final limiter filter for loud peaks (the normal Final Limiter is loudness preserving, but it doesnt handle very loud sounds well. This pre-final limiter filter removes those very loud sounds.) Loudness filter: The Loudness filter can now handle much louder peaks than before. This even makes it possible to use the Loudness filter as a limiter. Loudness filter: Loud S sounds now cause far less bass level reduction - so the sound is more stable. Loudness filter: Added a filter that detects and removes high frequency distortion caused by loud deep bass sounds. Loudness filter: Improved the filtering so it sounds smoother now. And with the same settings the sound is slightly louder than before with less artifacts. RDS: Added Now playing support for Winamp plugins without the need to use external files. RDS: Added date and time displays. Improved the HARD LIMIT filter, sound should more consistent (less pumping) now. Included the Singleband Compressor from versions 2.11 and below again. Its available from the Limiting Loudness window. Added some presets made by Vamprecords, SuperH and Bojcha. Added a Bypass all button on the main window. Added FM RDS TA button on the main window so traffic announcements can now be indicated without opening the FM configuration window. Performance: On multicore systems, the processing now uses multiple threads. Some slower AMD dualcore PCs couldnt handle FM output in High quality, now they can. Fixed a bug: The highpass filter seems to have been disabled when Multiband compression is turned on since version 3.50. Fixed a bug that could cause blocks of bad pixels in the user interface. Fixed a bug that caused presets to sound slightly different when they were selected while audio was playing. Fixed a bug: On startup, even if Multiband HQ was ticked Multiband didnt run in HQ mode. Greatly reduced the gaps that occur when switching between presets. Uncoupled screen redraws from processing. This allows Windows to select a different CPU for screen updates on multicore systems, and they might get disabled temporarily on single CPU systems when the load is high. Loudness filter: Louder and more stable bass sound. Soundcard support (stand alone version, Direct Soundcard Output on plugin versions): No more stuttering when the output buffer is empty, instead a short break is inserted such that the buffer is filled again when playback resumes. (No change for ASIO output). Loudness filter: Warmer, louder and more stable sound with deeper bass and cleaner highs. Added wave input and output displays. Added MPX spectrogram display for FM output. Fixed a bug that caused the Loudness filter in Normal quality mode to sound softer ( 0.3 dB for Loudness 4.0). FM transmitter bug fix: In version 3.52 the improved filtering caused RDS reception problems on some (mainly German) FM receivers. Fixed. Stereo widener is now almost as strong as in versions before 3.51, but with far less artifacts than before. Especially low quality MP3s should sound a lot better. FM transmitters: MPX filtering cleaned up further, noise level is now around 90 dB below the actual signal (used to be about 30 dB). Very minor change in pre-filtering steps that should slightly improve the handling of loud high frequencies. FM transmitters: Improved filtering for a cleaner MPX signal. Stereo widener causes less artifacts, but also has a bit less effect. Loudness filter: Recalibrated to achieve a much cleaner and louder sound. Distortion that was still present in version 3.40 is now nearly gone, and values up to 3.00 sound good. Values up to 4.00 are now allowed. Loud highs now have far less effect on the loudness of the other sounds - so it sounds much more stable. Added a filter to improve trumpet sounds. Improved quality and loudness of deep bass sounds. Improved FM pre-emphasis to match the official specification (in older versions the pre-emphasis wasnt strong enough, so this new version sounds brighter.) Volume slider now show both pre-emphasized and de-emphasized output levels. Changed Multiband compressor to get a warmer output sound with lower peak levels. Improved the HARD LIMIT filter to sound cleaner, cause far less pumping, and to achieve a louder output volume. Hard Limit filter is now also used if Direct Sound Card Output is used. Improved the composite limiter filter to cause far less pumping, and to achieve a louder and cleaner output volume. Updated presets, added FM Extreme preset. Improved Lowpass filter to filter closer to the selected lowpass frequency (important for AM presets). Also the filtering quality is now much better - the spectrum above the filtering frequency is now really empty. Bug fix: Fake stereo didnt work in version 3.40 - it does work again now. The CPU load is about 5-10 higher than that of version 3.40 - measured using the FM Loud preset. Loudness filter: Completely rewritten. Now reaches much louder sounds (up to 12 dB) with far less effects on the sound quality. Even notoriously difficult sounds such as xylophones are now handled without introducing distortion. Added FM transmitter calibration. If you are having problems with your FM signal (caused by low quality transmitter equipment, sound card, long cables etc.), you can now configure Stereo Tool to accomodate for that. (On my system, stereo separation is increased by 15 dB after calibrating) Composite limiter now operates at 705.6 or 768 kHz. This removes peaks that were introduced by the lowpass filter of the sound card. Hard limit now operates at 352.8 or 384 kHz. This removes peaks that were introduced by the lowpass filter of the sound card. Removed the Loudness option because it is no longer needed. Tweaked the presets to take advantage of the improved Loudness filter (more punch). The CPU load is roughly identical to that of version 3.30 in Low and Normal quality mode as long as the composite limiter oversampling is not used. And it will be a lot slower on PCs without SSE2 support. Composite limiter added. Gives about 7 (0.5 dB) extra loudness on FM transmitters, without affecting the FM Pilot and RDS signals. . Memory footprint reduced by about 12 MB (3.29 used 34 MB, 3.30 uses 22 MB)
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