我以不同的方式使用音訊製作了幾個項目： MIDI 檔案,語音API ，呼叫 Windows DLL，以及讀取/寫入 WAV 文件。
這次我將再次使用 WAV 檔案。 JMP 的 BlobToMatrix 和 MatrixToBlob 函數可對 blob（二進位大型物件）進行操作。 LoadTextFile 和 SaveTextFile 知道如何讀取和寫入 blob。 WAV 檔案是二進位物件（在文字編輯器中開啟 WAV 檔案不會帶來太多樂趣）。 JSL 矩陣中二進位位元組的記憶體表示形式為 8 個位元組長，這表示 1GB（1.5 小時 16 位元 44.1KHz 立體聲）WAV 檔案需要 8GB 記憶體。 對於我需要的 60 秒文件來說，這不是問題。同樣的想法可以用於任何中等大小的二進位文件，而不僅僅是 WAV，但正如您將看到的，解碼二進位檔案很複雜。
在之前的一個郵政我從網誌中收集了數據；我正在使用來自的 IP 到位置文件最大思維將 IP 位址轉換為位置和地名。 透過跨時間切片資料（一個月長的日誌檔案），我可以製作影片。 這是顯示一幀的圖形產生器。 我會製作 1800（30FPS * 60 秒）。
Color indicates five lat-lon clustersColor indicates five lat-lon clustersColor indicates five lat-lon clustersColor indicates five lat-lon clusters顏色表示五個經緯度簇
JMP 的聚類平台之一使用緯度和經度將點分為五個聚類；每個集群中的活動將驅動音樂的某些方面與視訊配合。 音樂生成演算法使用狀態機是由這個想法。 和弦過渡由其中一個顏色組驅動，打擊樂部分由另一個顏色組、吉他和低音吉他驅動。 可能我沒有使用全部五種顏色；當我聽到一些聽起來不錯的東西時，我就停下來了。 隨機不是一個好聲音。
樂器聲音（合唱團、吉他、打擊樂器）是從我製作的 WAV 檔案樣本中加載的LMMS 。 經過幾次錯誤的啟動後，我決定將樣本加載到 JMP 中的最簡單方法是將一種樂器的一組完整音符放入一個文件中，均勻間隔，音符之間有一些死區時間。
Really low guitar notes at the start of the WAV fileReally low guitar notes at the start of the WAV fileReally low guitar notes at the start of the WAV fileReally low guitar notes at the start of the WAV fileWAV 檔案開頭的吉他音符非常低
合唱團相同，打擊樂也相似。 儲存這些文件，然後將它們加載到 JMP 中，如下所示：
// RIFF loader
RiffUtil = New Namespace(
"RiffUtil"
);
這將是一個大型項目，因此從命名空間開始，以幫助將細節與項目的其他部分隔離。 RIFF 原來是 WAV 檔案背後的檔案格式的名稱。 在該名稱空間中將有幾個實用函數。 接下來，新增一個載入 WAV 的函數。
RiffUtil:loadWav = Function( {filename}, {Default Local},
riff = RiffUtil:load( filename );
{fmtOffset, fmtLength} = riff:chunks["fmt "];
riff:offset = fmtOffset;
// http://soundfile.sapp.org/doc/WaveFormat/
//JMP offset is 1-based, these are 0-based
//20 2 AudioFormat PCM = 1 (i.e. Linear quantization)
// Values other than 1 indicate some
// form of compression.
q = RiffUtil:getShort( riff, 1 );
riff:AudioFormat = q[1];
If( riff:AudioFormat != 1, Throw( "not PCM audio" ) );
//22 2 NumChannels Mono = 1, Stereo = 2, etc.
q = RiffUtil:getShort( riff, 1 );
riff:NumChannels = q[1];
If( riff:NumChannels != 2, Throw( "not 2-channel stereo" ) );
//24 4 SampleRate 8000, 44100, etc.
q = RiffUtil:getLong( riff, 1 );
riff:SampleRate = q[1];
If( riff:SampleRate != 44100, Throw( "not 44.1 kHz" ) );
//28 4 ByteRate == SampleRate * NumChannels * BitsPerSample/8
q = RiffUtil:getLong( riff, 1 );
riff:ByteRate = q[1];
If( riff:ByteRate != 44100 * 2 * 2, Throw( "not expected byte rate" ) );
//32 2 BlockAlign == NumChannels * BitsPerSample/8
// The number of bytes for one sample including
// all channels. I wonder what happens when
// this number isn't an integer?
q = RiffUtil:getShort( riff, 1 );
riff:BlockAlign = q[1];
If( riff:BlockAlign != 2 * 2, Throw( "not expected block align" ) );
//34 2 BitsPerSample 8 bits = 8, 16 bits = 16, etc.
q = RiffUtil:getShort( riff, 1 );
riff:BitsPerSample = q[1];
If( riff:BitsPerSample != 16, Throw( "not expected bits per sample" ) );
Show( riff:AudioFormat, riff:NumChannels, riff:SampleRate, riff:ByteRate, riff:BlockAlign, riff:BitsPerSample );
{dataOffset, dataLength} = riff:chunks["data"];
riff:offset = dataOffset;
nShorts = dataLength / 2;
result = Shape( RiffUtil:getShort( riff, nShorts ), nShorts / 2, 2 );
riff << delete;// the namespace
result; // return integers or... /32768;// return normalized +/- 1.0
);
大多數評論取自參考站點，並描述如何解析 WAV 檔案。 此程式碼處理一種 WAV 格式，如果格式錯誤則拋出例外。 （我只需要 16 位元 44.1KHz 立體聲。）當然，這個函數在執行其他操作之前會呼叫另一個載入函數。
RiffUtil:load = Function( {filename}, {Default Local},
riff = New Namespace();
riff:data = Blob To Matrix( Load Text File( filename, BLOB ), "uint", 1, "big" );
riff:offset = 1;
If( RiffUtil:getString( riff ) != "RIFF", Throw( filename || " is not a valid wav (RIFF missing)" ) );
q = RiffUtil:getLong( riff, 1 );
riff:length = q[1];
If( RiffUtil:getString( riff ) != "WAVE", Throw( filename || " is not a valid wav (WAVE missing)" ) );
// walk the chunks and record their offsets
riff:chunks = Associative Array();
While( riff:offset < riff:length - 8,
chunkName = RiffUtil:getString( riff );
q = RiffUtil:GetLong( riff, 1 );
chunkLength = q[1];
riff:chunks[chunkName] = Eval List( {riff:offset, chunkLength} );
riff:offset += chunkLength;
);
riff;//return
);
此函數完成繁重工作的第一部分，將 blob 載入到記憶體中，驗證 RIFF 標籤，然後驗證 WAVE 標籤，然後查看 WAVE 資料內部以分割指定的子區塊。 然後，早期的函數會對 fmt 和資料塊執行更繁重的工作。
RiffUtil:getString = Function( {riff},
{Default Local}, // return a 4-character id
four = riff:data[riff:offset + 0 :: riff:offset + 3];
riff:offset += 4;
Blob To Char( Matrix To Blob( four, "uint", 1, "big" ) );
);
getString 檢索 4 個位元組作為 JSL 字串；其他函數取得 1-2 或 4 位元組數字。 riff:offset 是一個變量，用於追蹤 blob 中的當前位置以供下一個要處理的位元組使用。 請注意上面的“四”是如何分配四個元素的矩陣，然後轉換為 blob，然後轉換為字串。 這是在沒有循環的情況下完成的並且運行速度很快。 這是數位版本。
// these two transforms assist in the two's complement conversion; if the MSB is on
// ("negative") then also subtract maximum value
RiffUtil:transform2 = ((0 :: 127) * (256 ^ 1)) || ((128 :: 255) * (256 ^ 1) - (256 ^ 2));
RiffUtil:getShort = Function( {riff, n},
{Default Local},
chooser = riff:offset + ((0 :: n - 1) * 2);
little = riff:data[chooser];
biggest = riff:data[chooser + 1];
riff:offset += 2 * n;
little + RiffUtil:transform2[biggest + 1];
);
RiffUtil:transform4 = ((0 :: 127) * (256 ^ 3)) || ((128 :: 255) * (256 ^ 3) - (256 ^ 4));
RiffUtil:getLong = Function( {riff, n},
{Default Local},
chooser = riff:offset + ((0 :: n - 1) * 4);
little = riff:data[chooser];
big = riff:data[chooser + 1];
bigger = riff:data[chooser + 2];
biggest = riff:data[chooser + 3];
riff:offset += 4 * n;
little + big * (256) + bigger * (256 ^ 2) + RiffUtil:transform4[biggest + 1];
);
二進制補碼是另一個主題。 稍後我們需要一個 wavWriter；這裡是。 它對於字串和數字具有完全相反的功能。
// wav writer...44.1 kHz 16 bit stereo
RiffUtil:WavWriter = Function( {filename, mat},
{Default Local},
uints = N Rows( mat ) * N Cols( mat );
wav = New Namespace();
wav:mat = J( 44 + 2 * uints, 1, 0 ); // pre-allocate buffer
wav:offset = 1; // 1-based
// http://soundfile.sapp.org/doc/WaveFormat/
//JMP offset is 1-based, these are 0-based
//0 4 ChunkID Contains the letters "RIFF" in ASCII form
// (0x52494646 big-endian form).
RiffUtil:putString( wav, "RIFF" );
//4 4 ChunkSize 36 + SubChunk2Size, or more precisely:
// 4 + (8 + SubChunk1Size) + (8 + SubChunk2Size)
// This is the size of the rest of the chunk
// following this number. This is the size of the
// entire file in bytes minus 8 bytes for the
// two fields not included in this count:
// ChunkID and ChunkSize.
RiffUtil:putLong( wav, J( 1, 1, 36 + 2 * uints ) );
//8 4 Format Contains the letters "WAVE"
// (0x57415645 big-endian form).
RiffUtil:putString( wav, "WAVE" );
//
//The "WAVE" format consists of two subchunks: "fmt " and "data":
//The "fmt " subchunk describes the sound data's format:
//
//12 4 Subchunk1ID Contains the letters "fmt "
// (0x666d7420 big-endian form).
RiffUtil:putString( wav, "fmt " );
//16 4 Subchunk1Size 16 for PCM. This is the size of the
// rest of the Subchunk which follows this number.
RiffUtil:putLong( wav, [16] );
//20 2 AudioFormat PCM = 1 (i.e. Linear quantization)
// Values other than 1 indicate some
// form of compression.
RiffUtil:putShort( wav, [1] );
//22 2 NumChannels Mono = 1, Stereo = 2, etc.
RiffUtil:putShort( wav, [2] );
//24 4 SampleRate 8000, 44100, etc.
RiffUtil:putLong( wav, [44100] );
//28 4 ByteRate == SampleRate * NumChannels * BitsPerSample/8
RiffUtil:putLong( wav, [176400] );
//32 2 BlockAlign == NumChannels * BitsPerSample/8
// The number of bytes for one sample including
// all channels. I wonder what happens when
// this number isn't an integer?
RiffUtil:putShort( wav, [4] );
//34 2 BitsPerSample 8 bits = 8, 16 bits = 16, etc.
RiffUtil:putShort( wav, [16] );
// 2 ExtraParamSize if PCM, then doesn't exist
// X ExtraParams space for extra parameters
//
//The "data" subchunk contains the size of the data and the actual sound:
//
//36 4 Subchunk2ID Contains the letters "data"
// (0x64617461 big-endian form).
RiffUtil:putString( wav, "data" );
//40 4 Subchunk2Size == NumSamples * NumChannels * BitsPerSample/8
// This is the number of bytes in the data.
// You can also think of this as the size
// of the read of the subchunk following this
// number.
RiffUtil:putLong( wav, J( 1, 1, 2 * uints ) );
//44 * Data The actual sound data.
RiffUtil:putShort( wav, mat );
Open( Save Text File( filename, Matrix To Blob( wav:mat, "uint", 1, "big" ) ) );
wav << delete;
);
RiffUtil:putString = Function( {wav, txt},
{Default Local},
mat = Blob To Matrix( Char To Blob( txt ), "uint", 1, "big" );
len = N Rows( mat );
wav:mat[wav:offset + 0 :: wav:offset + len - 1] = mat;
wav:offset += len;
0;//return
);
// should longs in wav files ever go negative? what about > 2GB data blocks?
RiffUtil:putLong = Function( {wav, mat},
{Default Local},
//show(mat);
len = N Rows( mat ) * N Cols( mat );
biggest = Floor( mat / 256 ^ 3 );
mat -= biggest * 256 ^ 3;
bigger = Floor( mat / 256 ^ 2 );
mat -= bigger * 256 ^ 2;
big = Floor( mat / 256 );
mat -= big * 256;
little = mat;
wav:mat[wav:offset + 0 :: wav:offset + 4 * len - 1] = Shape( little || big || bigger || biggest, 1, 4 * len );
wav:offset += 4 * len;
);
RiffUtil:putShort = Function( {wav, mat},
{Default Local}, // mat has two stereo columns of -32k...+32k ints
mat[Loc( mat < 0 )] += 256 ^ 2; // two's complement: -1==65535 ... -32768 == 32768
len = N Rows( mat ) * N Cols( mat ); // len unsigned (or two's complement) 16-bit values (L+R)
mat = Shape( mat, len, 1 ); // now mat is a list of 16 bit samples, one per row
big = Floor( mat / 256 );
mat -= big * 256; // high bytes
little = mat; // low bytes
//show(nrows(mat),ncols(mat),wav:offset,len,nrows(wav:mat),ncols(wav:mat),nrows(little),ncols(little));
wav:mat[wav:offset + 0 :: wav:offset + 2 * len - 1] = Shape( little || big, 1, 2 * len );
wav:offset += 2 * len;
);
您可能已經注意到 riff<<
現在是時候加載筆記了。 我將為筆記載入器 JSL 建立另一個名稱空間以保持其隔離。
noteLoader = New Namespace(
"noteLoader"
);
noteLoader:load1 = Function( {path, samples, firstnote},
{Default Local},
Show( path );
result = New Namespace();
result:notes = Associative Array();
temp = RiffUtil:loadWav( path );
For( pos = 1, pos < N Rows( temp ) - samples, pos += samples,
//show(firstnote,pos,pos+samples-1,nrows(temp),ncols(temp));
result:notes[firstnote] = temp[pos :: pos + samples - 1, 0];
firstnote += 1;
);
result; // return
);
「load1」完成了我為這個專案確定的事情；隨著專案的進展，「負載」逐漸消失。 load1 使用 wav 載入器取得音訊資料矩陣，然後使用樣本參數逐步遍歷資料並為單一音符建立較短的音訊資料矩陣的關聯數組。 Firstnote 是（由我）分配給音符升序序列的數字。
還有一個作曲家名稱空間；作曲家 JSL 知道如何在樂曲中加入註解。
composer = New Namespace(
"composer"
);
composer:create = Function( {seconds},
{Default Local},
composer:mat = J( seconds * 44100, 2, 0 );
composer:choirPad4 = noteLoader:load1( path || "choir1/choirPad4.wav", 44100 * 8, 36);
composer:guitarVarA = noteLoader:load1( path || "Guitar1/varA.wav", 44100 * 2, 12 );
composer:guitarVarB = noteLoader:load1( path || "Guitar1/varB.wav", 44100 * 2, 12 );
composer:guitarVarC = noteLoader:load1( path || "Guitar1/varC.wav", 44100 * 2, 12 );
composer:guitarVarD = noteLoader:load1( path || "Guitar1/varD.wav", 44100 * 2, 12 );
composer:percus = noteLoader:load1( path || "Drum1/percus.wav", 44100 * 2, 1 );
);
composer:compose = Function( {instrument, note, position, volume}, {Default Local},
position = Round( 1 + position * 44100 );
duration = N Rows( instrument:notes[note] );
composer:mat[position :: (position + duration - 1), 0] += (instrument:notes[note]) * volume;
);
composer:new = Function( {}, {Default Local},
composer:mat[0, 0] = 0
);
composer:limit = Function( {}, {},
N Rows( composer:mat )
);
composer:save = Function( {filename}, {Default Local},
x = Max( Abs( composer:mat ) );
composer:mat = (20000 / x) * composer:mat; // normalize to +/- 20,000 out of +/- 32767
RiffUtil:wavWriter( filename, composer:mat );
);
作曲家也知道如何重新開始（新）和完成（保存）。
這是 YouTube 影片中的音訊。 如果您獲得 LMMS 的副本來製作 WAV 文件，則附加文件將大致複製它（由於隨機選擇）。 載入 LMMS 專案並匯出。
