如何使用 JS WebAudioAPI 进行节拍检测?
How can I use JS WebAudioAPI for beat detection?
我有兴趣使用 JavaScript WebAudioAPI
来检测歌曲节拍,然后在 canvas 中呈现它们。
我可以处理 canvas 部分,但我不是音频专家,真的不明白如何在 JavaScript 中制作节拍检测器。
我试过遵循 this article 但我无法将每个函数之间的点连接起来以制作功能程序。
我知道我应该给你看一些代码,但老实说我没有,我所有的尝试都惨败,相关代码在前面提到的文章中。
无论如何,我非常感谢一些指导,或者更好的演示如何使用 WebAudioAPI
.
实际检测歌曲节拍
谢谢!
关于the referenced article by Joe Sullivan的主要理解是,尽管它提供了大量源代码,但它远非最终和完整的代码。要获得可行的解决方案,您仍然需要一些编码和调试技能。
此答案的大部分代码来自参考文章,原始许可适用于适当的地方。
以下是使用上述文章中描述的功能的简单示例实现,您仍然需要找出功能解决方案的正确阈值。
代码由为答案编写的准备代码组成:
- 正在通过 the FileReader API
读取本地文件
- 使用the AudioContext API
将文件解码为音频数据
然后,如文章所述:
- 过滤音频,在本例中使用 low-pass filter
- 使用阈值计算峰值
- 分组间隔计数,然后分组节奏计数
对于阈值,我使用了最大值和最小值之间范围的任意值 .98;分组时我添加了一些额外的检查和任意舍入以避免可能的无限循环并使其成为易于调试的示例。
请注意,为了使示例实现简短,注释很少,因为:
- 参考文章中解释了处理背后的逻辑
- 语法可以参考相关方法的API文档
audio_file.onchange = function() {
var file = this.files[0];
var reader = new FileReader();
var context = new(window.AudioContext || window.webkitAudioContext)();
reader.onload = function() {
context.decodeAudioData(reader.result, function(buffer) {
prepare(buffer);
});
};
reader.readAsArrayBuffer(file);
};
function prepare(buffer) {
var offlineContext = new OfflineAudioContext(1, buffer.length, buffer.sampleRate);
var source = offlineContext.createBufferSource();
source.buffer = buffer;
var filter = offlineContext.createBiquadFilter();
filter.type = "lowpass";
source.connect(filter);
filter.connect(offlineContext.destination);
source.start(0);
offlineContext.startRendering();
offlineContext.oncomplete = function(e) {
process(e);
};
}
function process(e) {
var filteredBuffer = e.renderedBuffer;
//If you want to analyze both channels, use the other channel later
var data = filteredBuffer.getChannelData(0);
var max = arrayMax(data);
var min = arrayMin(data);
var threshold = min + (max - min) * 0.98;
var peaks = getPeaksAtThreshold(data, threshold);
var intervalCounts = countIntervalsBetweenNearbyPeaks(peaks);
var tempoCounts = groupNeighborsByTempo(intervalCounts);
tempoCounts.sort(function(a, b) {
return b.count - a.count;
});
if (tempoCounts.length) {
output.innerHTML = tempoCounts[0].tempo;
}
}
// http://tech.beatport.com/2014/web-audio/beat-detection-using-web-audio/
function getPeaksAtThreshold(data, threshold) {
var peaksArray = [];
var length = data.length;
for (var i = 0; i < length;) {
if (data[i] > threshold) {
peaksArray.push(i);
// Skip forward ~ 1/4s to get past this peak.
i += 10000;
}
i++;
}
return peaksArray;
}
function countIntervalsBetweenNearbyPeaks(peaks) {
var intervalCounts = [];
peaks.forEach(function(peak, index) {
for (var i = 0; i < 10; i++) {
var interval = peaks[index + i] - peak;
var foundInterval = intervalCounts.some(function(intervalCount) {
if (intervalCount.interval === interval) return intervalCount.count++;
});
//Additional checks to avoid infinite loops in later processing
if (!isNaN(interval) && interval !== 0 && !foundInterval) {
intervalCounts.push({
interval: interval,
count: 1
});
}
}
});
return intervalCounts;
}
function groupNeighborsByTempo(intervalCounts) {
var tempoCounts = [];
intervalCounts.forEach(function(intervalCount) {
//Convert an interval to tempo
var theoreticalTempo = 60 / (intervalCount.interval / 44100);
theoreticalTempo = Math.round(theoreticalTempo);
if (theoreticalTempo === 0) {
return;
}
// Adjust the tempo to fit within the 90-180 BPM range
while (theoreticalTempo < 90) theoreticalTempo *= 2;
while (theoreticalTempo > 180) theoreticalTempo /= 2;
var foundTempo = tempoCounts.some(function(tempoCount) {
if (tempoCount.tempo === theoreticalTempo) return tempoCount.count += intervalCount.count;
});
if (!foundTempo) {
tempoCounts.push({
tempo: theoreticalTempo,
count: intervalCount.count
});
}
});
return tempoCounts;
}
//
function arrayMin(arr) {
var len = arr.length,
min = Infinity;
while (len--) {
if (arr[len] < min) {
min = arr[len];
}
}
return min;
}
function arrayMax(arr) {
var len = arr.length,
max = -Infinity;
while (len--) {
if (arr[len] > max) {
max = arr[len];
}
}
return max;
}
<input id="audio_file" type="file" accept="audio/*"></input>
<audio id="audio_player"></audio>
<p>
Most likely tempo: <span id="output"></span>
</p>
我在这里写了一个教程,展示了如何使用 javascript 网络音频 API。
https://askmacgyver.com/blog/tutorial/how-to-implement-tempo-detection-in-your-application
步骤大纲
- 将音频文件转换为数组缓冲区
- 运行 通过低通滤波器的数组缓冲区
- Trim 来自数组缓冲区的 10 秒剪辑
- 对数据进行下采样
- 标准化数据
- 计数卷分组
- 根据分组计数推断节奏
下面的这段代码完成了繁重的工作。
将音频文件加载到数组缓冲区并运行通过低通滤波器
function createBuffers(url) {
// Fetch Audio Track via AJAX with URL
request = new XMLHttpRequest();
request.open('GET', url, true);
request.responseType = 'arraybuffer';
request.onload = function(ajaxResponseBuffer) {
// Create and Save Original Buffer Audio Context in 'originalBuffer'
var audioCtx = new AudioContext();
var songLength = ajaxResponseBuffer.total;
// Arguments: Channels, Length, Sample Rate
var offlineCtx = new OfflineAudioContext(1, songLength, 44100);
source = offlineCtx.createBufferSource();
var audioData = request.response;
audioCtx.decodeAudioData(audioData, function(buffer) {
window.originalBuffer = buffer.getChannelData(0);
var source = offlineCtx.createBufferSource();
source.buffer = buffer;
// Create a Low Pass Filter to Isolate Low End Beat
var filter = offlineCtx.createBiquadFilter();
filter.type = "lowpass";
filter.frequency.value = 140;
source.connect(filter);
filter.connect(offlineCtx.destination);
// Render this low pass filter data to new Audio Context and Save in 'lowPassBuffer'
offlineCtx.startRendering().then(function(lowPassAudioBuffer) {
var audioCtx = new(window.AudioContext || window.webkitAudioContext)();
var song = audioCtx.createBufferSource();
song.buffer = lowPassAudioBuffer;
song.connect(audioCtx.destination);
// Save lowPassBuffer in Global Array
window.lowPassBuffer = song.buffer.getChannelData(0);
console.log("Low Pass Buffer Rendered!");
});
},
function(e) {});
}
request.send();
}
createBuffers('https://askmacgyver.com/test/Maroon5-Moves-Like-Jagger-128bpm.mp3');
您现在拥有低通滤波歌曲(和原始歌曲)的数组缓冲区
它由多个条目组成,sampleRate(44100 乘以歌曲的秒数)。
window.lowPassBuffer // Low Pass Array Buffer
window.originalBuffer // Original Non Filtered Array Buffer
Trim 歌曲中的 10 秒剪辑
function getClip(length, startTime, data) {
var clip_length = length * 44100;
var section = startTime * 44100;
var newArr = [];
for (var i = 0; i < clip_length; i++) {
newArr.push(data[section + i]);
}
return newArr;
}
// Overwrite our array buffer to a 10 second clip starting from 00:10s
window.lowPassFilter = getClip(10, 10, lowPassFilter);
对剪辑进行降采样
function getSampleClip(data, samples) {
var newArray = [];
var modulus_coefficient = Math.round(data.length / samples);
for (var i = 0; i < data.length; i++) {
if (i % modulus_coefficient == 0) {
newArray.push(data[i]);
}
}
return newArray;
}
// Overwrite our array to down-sampled array.
lowPassBuffer = getSampleClip(lowPassFilter, 300);
标准化您的数据
function normalizeArray(data) {
var newArray = [];
for (var i = 0; i < data.length; i++) {
newArray.push(Math.abs(Math.round((data[i + 1] - data[i]) * 1000)));
}
return newArray;
}
// Overwrite our array to the normalized array
lowPassBuffer = normalizeArray(lowPassBuffer);
平线分组数
function countFlatLineGroupings(data) {
var groupings = 0;
var newArray = normalizeArray(data);
function getMax(a) {
var m = -Infinity,
i = 0,
n = a.length;
for (; i != n; ++i) {
if (a[i] > m) {
m = a[i];
}
}
return m;
}
function getMin(a) {
var m = Infinity,
i = 0,
n = a.length;
for (; i != n; ++i) {
if (a[i] < m) {
m = a[i];
}
}
return m;
}
var max = getMax(newArray);
var min = getMin(newArray);
var count = 0;
var threshold = Math.round((max - min) * 0.2);
for (var i = 0; i < newArray.length; i++) {
if (newArray[i] > threshold && newArray[i + 1] < threshold && newArray[i + 2] < threshold && newArray[i + 3] < threshold && newArray[i + 6] < threshold) {
count++;
}
}
return count;
}
// Count the Groupings
countFlatLineGroupings(lowPassBuffer);
将 10 秒分组计数扩展到 60 秒以得出每分钟节拍
var final_tempo = countFlatLineGroupings(lowPassBuffer);
// final_tempo will be 21
final_tempo = final_tempo * 6;
console.log("Tempo: " + final_tempo);
// final_tempo will be 126
我有兴趣使用 JavaScript WebAudioAPI
来检测歌曲节拍,然后在 canvas 中呈现它们。
我可以处理 canvas 部分,但我不是音频专家,真的不明白如何在 JavaScript 中制作节拍检测器。
我试过遵循 this article 但我无法将每个函数之间的点连接起来以制作功能程序。
我知道我应该给你看一些代码,但老实说我没有,我所有的尝试都惨败,相关代码在前面提到的文章中。
无论如何,我非常感谢一些指导,或者更好的演示如何使用 WebAudioAPI
.
谢谢!
关于the referenced article by Joe Sullivan的主要理解是,尽管它提供了大量源代码,但它远非最终和完整的代码。要获得可行的解决方案,您仍然需要一些编码和调试技能。
此答案的大部分代码来自参考文章,原始许可适用于适当的地方。
以下是使用上述文章中描述的功能的简单示例实现,您仍然需要找出功能解决方案的正确阈值。
代码由为答案编写的准备代码组成:
- 正在通过 the FileReader API 读取本地文件
- 使用the AudioContext API 将文件解码为音频数据
然后,如文章所述:
- 过滤音频,在本例中使用 low-pass filter
- 使用阈值计算峰值
- 分组间隔计数,然后分组节奏计数
对于阈值,我使用了最大值和最小值之间范围的任意值 .98;分组时我添加了一些额外的检查和任意舍入以避免可能的无限循环并使其成为易于调试的示例。
请注意,为了使示例实现简短,注释很少,因为:
- 参考文章中解释了处理背后的逻辑
- 语法可以参考相关方法的API文档
audio_file.onchange = function() {
var file = this.files[0];
var reader = new FileReader();
var context = new(window.AudioContext || window.webkitAudioContext)();
reader.onload = function() {
context.decodeAudioData(reader.result, function(buffer) {
prepare(buffer);
});
};
reader.readAsArrayBuffer(file);
};
function prepare(buffer) {
var offlineContext = new OfflineAudioContext(1, buffer.length, buffer.sampleRate);
var source = offlineContext.createBufferSource();
source.buffer = buffer;
var filter = offlineContext.createBiquadFilter();
filter.type = "lowpass";
source.connect(filter);
filter.connect(offlineContext.destination);
source.start(0);
offlineContext.startRendering();
offlineContext.oncomplete = function(e) {
process(e);
};
}
function process(e) {
var filteredBuffer = e.renderedBuffer;
//If you want to analyze both channels, use the other channel later
var data = filteredBuffer.getChannelData(0);
var max = arrayMax(data);
var min = arrayMin(data);
var threshold = min + (max - min) * 0.98;
var peaks = getPeaksAtThreshold(data, threshold);
var intervalCounts = countIntervalsBetweenNearbyPeaks(peaks);
var tempoCounts = groupNeighborsByTempo(intervalCounts);
tempoCounts.sort(function(a, b) {
return b.count - a.count;
});
if (tempoCounts.length) {
output.innerHTML = tempoCounts[0].tempo;
}
}
// http://tech.beatport.com/2014/web-audio/beat-detection-using-web-audio/
function getPeaksAtThreshold(data, threshold) {
var peaksArray = [];
var length = data.length;
for (var i = 0; i < length;) {
if (data[i] > threshold) {
peaksArray.push(i);
// Skip forward ~ 1/4s to get past this peak.
i += 10000;
}
i++;
}
return peaksArray;
}
function countIntervalsBetweenNearbyPeaks(peaks) {
var intervalCounts = [];
peaks.forEach(function(peak, index) {
for (var i = 0; i < 10; i++) {
var interval = peaks[index + i] - peak;
var foundInterval = intervalCounts.some(function(intervalCount) {
if (intervalCount.interval === interval) return intervalCount.count++;
});
//Additional checks to avoid infinite loops in later processing
if (!isNaN(interval) && interval !== 0 && !foundInterval) {
intervalCounts.push({
interval: interval,
count: 1
});
}
}
});
return intervalCounts;
}
function groupNeighborsByTempo(intervalCounts) {
var tempoCounts = [];
intervalCounts.forEach(function(intervalCount) {
//Convert an interval to tempo
var theoreticalTempo = 60 / (intervalCount.interval / 44100);
theoreticalTempo = Math.round(theoreticalTempo);
if (theoreticalTempo === 0) {
return;
}
// Adjust the tempo to fit within the 90-180 BPM range
while (theoreticalTempo < 90) theoreticalTempo *= 2;
while (theoreticalTempo > 180) theoreticalTempo /= 2;
var foundTempo = tempoCounts.some(function(tempoCount) {
if (tempoCount.tempo === theoreticalTempo) return tempoCount.count += intervalCount.count;
});
if (!foundTempo) {
tempoCounts.push({
tempo: theoreticalTempo,
count: intervalCount.count
});
}
});
return tempoCounts;
}
//
function arrayMin(arr) {
var len = arr.length,
min = Infinity;
while (len--) {
if (arr[len] < min) {
min = arr[len];
}
}
return min;
}
function arrayMax(arr) {
var len = arr.length,
max = -Infinity;
while (len--) {
if (arr[len] > max) {
max = arr[len];
}
}
return max;
}
<input id="audio_file" type="file" accept="audio/*"></input>
<audio id="audio_player"></audio>
<p>
Most likely tempo: <span id="output"></span>
</p>
我在这里写了一个教程,展示了如何使用 javascript 网络音频 API。
https://askmacgyver.com/blog/tutorial/how-to-implement-tempo-detection-in-your-application
步骤大纲
- 将音频文件转换为数组缓冲区
- 运行 通过低通滤波器的数组缓冲区
- Trim 来自数组缓冲区的 10 秒剪辑
- 对数据进行下采样
- 标准化数据
- 计数卷分组
- 根据分组计数推断节奏
下面的这段代码完成了繁重的工作。
将音频文件加载到数组缓冲区并运行通过低通滤波器
function createBuffers(url) {
// Fetch Audio Track via AJAX with URL
request = new XMLHttpRequest();
request.open('GET', url, true);
request.responseType = 'arraybuffer';
request.onload = function(ajaxResponseBuffer) {
// Create and Save Original Buffer Audio Context in 'originalBuffer'
var audioCtx = new AudioContext();
var songLength = ajaxResponseBuffer.total;
// Arguments: Channels, Length, Sample Rate
var offlineCtx = new OfflineAudioContext(1, songLength, 44100);
source = offlineCtx.createBufferSource();
var audioData = request.response;
audioCtx.decodeAudioData(audioData, function(buffer) {
window.originalBuffer = buffer.getChannelData(0);
var source = offlineCtx.createBufferSource();
source.buffer = buffer;
// Create a Low Pass Filter to Isolate Low End Beat
var filter = offlineCtx.createBiquadFilter();
filter.type = "lowpass";
filter.frequency.value = 140;
source.connect(filter);
filter.connect(offlineCtx.destination);
// Render this low pass filter data to new Audio Context and Save in 'lowPassBuffer'
offlineCtx.startRendering().then(function(lowPassAudioBuffer) {
var audioCtx = new(window.AudioContext || window.webkitAudioContext)();
var song = audioCtx.createBufferSource();
song.buffer = lowPassAudioBuffer;
song.connect(audioCtx.destination);
// Save lowPassBuffer in Global Array
window.lowPassBuffer = song.buffer.getChannelData(0);
console.log("Low Pass Buffer Rendered!");
});
},
function(e) {});
}
request.send();
}
createBuffers('https://askmacgyver.com/test/Maroon5-Moves-Like-Jagger-128bpm.mp3');
您现在拥有低通滤波歌曲(和原始歌曲)的数组缓冲区
它由多个条目组成,sampleRate(44100 乘以歌曲的秒数)。
window.lowPassBuffer // Low Pass Array Buffer
window.originalBuffer // Original Non Filtered Array Buffer
Trim 歌曲中的 10 秒剪辑
function getClip(length, startTime, data) {
var clip_length = length * 44100;
var section = startTime * 44100;
var newArr = [];
for (var i = 0; i < clip_length; i++) {
newArr.push(data[section + i]);
}
return newArr;
}
// Overwrite our array buffer to a 10 second clip starting from 00:10s
window.lowPassFilter = getClip(10, 10, lowPassFilter);
对剪辑进行降采样
function getSampleClip(data, samples) {
var newArray = [];
var modulus_coefficient = Math.round(data.length / samples);
for (var i = 0; i < data.length; i++) {
if (i % modulus_coefficient == 0) {
newArray.push(data[i]);
}
}
return newArray;
}
// Overwrite our array to down-sampled array.
lowPassBuffer = getSampleClip(lowPassFilter, 300);
标准化您的数据
function normalizeArray(data) {
var newArray = [];
for (var i = 0; i < data.length; i++) {
newArray.push(Math.abs(Math.round((data[i + 1] - data[i]) * 1000)));
}
return newArray;
}
// Overwrite our array to the normalized array
lowPassBuffer = normalizeArray(lowPassBuffer);
平线分组数
function countFlatLineGroupings(data) {
var groupings = 0;
var newArray = normalizeArray(data);
function getMax(a) {
var m = -Infinity,
i = 0,
n = a.length;
for (; i != n; ++i) {
if (a[i] > m) {
m = a[i];
}
}
return m;
}
function getMin(a) {
var m = Infinity,
i = 0,
n = a.length;
for (; i != n; ++i) {
if (a[i] < m) {
m = a[i];
}
}
return m;
}
var max = getMax(newArray);
var min = getMin(newArray);
var count = 0;
var threshold = Math.round((max - min) * 0.2);
for (var i = 0; i < newArray.length; i++) {
if (newArray[i] > threshold && newArray[i + 1] < threshold && newArray[i + 2] < threshold && newArray[i + 3] < threshold && newArray[i + 6] < threshold) {
count++;
}
}
return count;
}
// Count the Groupings
countFlatLineGroupings(lowPassBuffer);
将 10 秒分组计数扩展到 60 秒以得出每分钟节拍
var final_tempo = countFlatLineGroupings(lowPassBuffer);
// final_tempo will be 21
final_tempo = final_tempo * 6;
console.log("Tempo: " + final_tempo);
// final_tempo will be 126