Chapter 3. Volume and Loudness

Once we are ready to play a sound, whether from an AudioBuffer or from other sources, one of the most basic parameters we can change is the loudness of the sound.

The main way to affect the loudness of a sound is using GainNodes. As previously mentioned, these nodes have a gain parameter, which acts as a multiplier on the incoming sound buffer. The default gain value is one, which means that the input sound is unaffected. Values between zero and one reduce the loudness, and values greater than one amplify the loudness. Negative gain (values less than zero) inverts the waveform (i.e., the amplitude is flipped).

Equal Power Crossfading

Often in a game setting, you have a situation where you want to crossfade between two environments that have different sounds associated with them. However, when to crossfade and by how much is not known in advance; perhaps it varies with the position of the game avatar, which is controlled by the player. In this case, we cannot do an automatic ramp.

In general, doing a straightforward, linear fade will result in the following graph. It can sound unbalanced because of a volume dip between the two samples, as shown in Figure 3-2.

A linear crossfade between two tracks
Figure 3-2. A linear crossfade between two tracks

To address this issue, we use an equal power curve, in which the corresponding gain curves are neither linear nor exponential, and intersect at a higher amplitude (Figure 3-3). This helps avoid a dip in volume in the middle part of the crossfade, when both sounds are mixed together equally.

An equal power crossfade sounds much better
Figure 3-3. An equal power crossfade sounds much better

The graph in Figure 3-3 can be generated with a bit of math:

function equalPowerCrossfade(percent) {
  // Use an equal-power crossfading curve:
  var gain1 = Math.cos(percent * 0.5*Math.PI);
  var gain2 = Math.cos((1.0 - percent) * 0.5*Math.PI);
  this.ctl1.gainNode.gain.value = gain1;
  this.ctl2.gainNode.gain.value = gain2;

Using Meters to Detect and Prevent Clipping

Since multiple sounds playing simultaneously are additive with no level reduction, you may find yourself in a situation where you are exceeding past the threshold of your speaker’s capability. The maximum level of sound is 0 dBFS, or 216, for 16-bit audio. In the floating point version of the signal, these bit values are mapped to [−1, 1]. The waveform of a sound that’s being clipped looks something like Figure 3-5. In the context of the Web Audio API, sounds clip if the values sent to the destination node lie outside of the range. It’s a good idea to leave some room (called headroom) in your final mix so that you aren’t too close to the clipping threshold.

A diagram of a waveform being clipped
Figure 3-5. A diagram of a waveform being clipped

In addition to close listening, you can check whether or not you are clipping your sound programmatically by putting a script processor node into your audio graph. Clipping may occur if any of the PCM values are out of the acceptable range. In this sample, we check both left and right channels for clipping, and if clipping is detected, save the last clipping time:

function onProcess(e) {
  var leftBuffer = e.inputBuffer.getChannelData(0);
  var rightBuffer = e.inputBuffer.getChannelData(1);

function checkClipping(buffer) {
  var isClipping = false;
  // Iterate through buffer to check if any of the |values| exceeds 1.
  for (var i = 0; i < buffer.length; i++) {
    var absValue = Math.abs(buffer[i]);
    if (absValue >= 1.0) {
      isClipping = true;
  this.isClipping = isClipping;
  if (isClipping) {
    lastClipTime = new Date();

An alternative implementation of metering could poll a real-time analyzer in the audio graph for getFloatFrequencyData at render time, as determined by requestAnimationFrame (see Chapter 5). This approach is more efficient, but misses a lot of the signal (including places where it potentially clips), since rendering happens most at 60 times a second, whereas the audio signal changes far more quickly.

The way to prevent clipping is to reduce the overall level of the signal. If you are clipping, apply some fractional gain on a master audio gain node to subdue your mix to a level that prevents clipping. In general, you should tweak gains to anticipate the worst case, but getting this right is more of an art than a science. In practice, since the sounds playing in your game or interactive application may depend on a huge variety of factors that are decided at runtime, it can be difficult to pick the master gain value that prevents clipping in all cases. For this unpredictable case, look to dynamics compression, which is discussed in “Dynamics Compression”.

Dynamics Compression

Compressors are available in the Web Audio API as DynamicsCompressorNodes. Using moderate amounts of dynamics compression in your mix is generally a good idea, especially in a game setting where, as previously discussed, you don’t know exactly what sounds will play and when. One case where compression should be avoided is when dealing with painstakingly mastered tracks that have been tuned to sound “just right” already, which are not being mixed with any other tracks.

Implementing dynamic compression in the Web Audio API is simply a matter of including a dynamics compressor node in your audio graph, generally as the last node before the destination:

var compressor = context.createDynamicsCompressor();

The node can be configured with some additional parameters as described in the theory section, but the defaults are quite good for most purposes. For more information about configuring the compression curve, see the Web Audio API specification.