我以不同的方式使用音頻製作了幾個項目： midi文件,語音接口, 調用 Windows DLL, 和讀/寫 WAV 文件.
這次我將再次使用 WAV 文件。 JMP 的 BlobToMatrix 和 MatrixToBlob 函數可對 blob（二進制大對象）進行操作。 LoadTextFile 和 SaveTextFile 知道如何讀取和寫入 blob。 WAV 文件是一個二進制對象（在文本編輯器中打開 WAV 文件不會有太多樂趣）。 JSL 矩陣中二進製字節的內存表示形式為八字節長，這意味著 1GB（1.5 小時 16 位 44.1KHz 立體聲）WAV 文件需要 8GB 內存。 對於我需要的 60 秒文件，這不是問題。同樣的想法可以用於任何中等大小的二進製文件，而不僅僅是 WAV，但正如您將看到的，解碼二進製文件很複雜。
在之前的郵政我從網絡日誌中收集數據；我正在使用來自的 IP-to-location 文件極智將 IP 地址轉換為位置和地名。 通過跨時間切片數據（一個月長的日誌文件），我可以製作視頻。 這是顯示一幀的圖形生成器。 我將製作 1800（30FPS * 60 秒）。
Color indicates five lat-lon clusters顏色表示五個 lat-lon 集群
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頻一起使用。 音樂生成算法使用狀態機那是從這個想法. 和弦過渡由其中一個顏色組驅動，打擊樂器部分由另一個顏色組驅動，還有吉他和貝斯吉他。 可能我沒有使用所有五種顏色；當我有一些聽起來不錯的東西時，我就停下來了。 隨機不是好聲音。
樂器聲音（合唱、吉他、打擊樂）是從我製作的 WAV 文件樣本中加載的LMMS . 在幾次錯誤的開始之後，我決定將樣本加載到 JMP 中的最簡單方法是將一個儀器的一整套音符放在一個文件中，間隔均勻，音符之間有一些死時間。
Really low guitar notes at the start of the WAV fileWAV 文件開頭的吉他音符非常低
合唱團相同，打擊樂相似。 保存這些文件，然後像這樣將它們加載到 JMP 中：
// 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開來。 RIFF 原來是 WAV 文件背後的文件格式的名稱。 這個名稱空間中將有幾個實用函數。 接下來，添加一個函數來加載 WAV。
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，描述了 WAV 文件是如何解析的。 此代碼處理一種 WAV 格式並在格式錯誤時拋出。 （我只需要 16 位 44.1KHz 立體聲。）當然，此函數在執行任何其他操作之前會調用另一個加載函數。
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
);
此函數執行繁重工作的第一部分，將 blob 加載到內存中，驗證 RIFF 標籤，然後驗證 WAVE 標籤，然後查看 WAVE 數據內部以拆分出命名的子塊。 然後，較早的函數對 fmt 和數據塊進行更多繁重的工作。
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 個字節作為 JSL 字符串；其他函數得到 1-2 或 4 字節的數字。 riff:offset 是一個變量，用於跟踪要處理的下一個字節在 blob 中的當前位置。 請注意上面的“四”是如何分配給一個包含四個元素的矩陣，然後轉換為一個 blob，然後再轉換為一個字符串。 這是在沒有循環的情況下完成的並且運行速度很快。 這是數字版本。
// 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。 它對字符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;
);
您可能已經註意到 riff<<
現在是加載筆記的時候了。 我將為筆記加載器 JSL 創建另一個名稱空間以使其保持隔離。
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”做了我為這個項目確定的事情；隨著項目的進展，“負載”逐漸下降。 load1 使用 wav 加載器獲取音頻數據矩陣，然後使用 samples 參數遍歷數據並為單個音符創建一個較短音頻數據矩陣的關聯數組。 Firstnote 是（由我）分配給音符升序的數字。
還有一個作曲家名字空間；作曲家 JSL 知道如何在作品中添加音符。
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 );
);
作曲家還知道如何重新開始（新）和完成（保存）。
這是 youtube 視頻中的音頻。 如果您獲得 LMMS 的副本來製作 WAV 文件，則附件將大約（由於隨機選擇）複製它。 加載 LMMS 項目並導出。
