mirror of
https://github.com/arcodange-org/mediabunny.git
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Fix Conversion API process callbacks being called *before* other transformations (fixes #403), restructure Conversion API to make full use of in-source transformations, add AudioSample.trim()
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+10
-15
@@ -20,41 +20,32 @@ export class AudioResampler {
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targetSampleRate: number;
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sourceNumberOfChannels: number | null = null;
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targetNumberOfChannels: number;
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endTime: number;
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startTime: number | null = null;
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onSample: (sample: AudioSample) => Promise<void>;
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bufferSizeInFrames: number;
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bufferSizeInSamples: number;
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outputBuffer: Float32Array;
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/** Start frame of current buffer */
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bufferStartFrame: number;
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bufferStartFrame = 0;
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/** The highest index written to in the current buffer */
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maxWrittenFrame: number | null = null;
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channelMixer!: (sourceData: Float32Array, sourceFrameIndex: number, targetChannelIndex: number) => number;
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tempSourceBuffer!: Float32Array;
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timestampOffset: number;
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constructor(options: {
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targetSampleRate: number;
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targetNumberOfChannels: number;
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startTime: number;
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endTime: number;
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onSample: (sample: AudioSample) => Promise<void>;
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}) {
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this.targetSampleRate = options.targetSampleRate;
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this.targetNumberOfChannels = options.targetNumberOfChannels;
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this.endTime = options.endTime;
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this.onSample = options.onSample;
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this.bufferSizeInFrames = Math.floor(this.targetSampleRate * 5.0); // 5 seconds
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this.bufferSizeInSamples = this.bufferSizeInFrames * this.targetNumberOfChannels;
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this.outputBuffer = new Float32Array(this.bufferSizeInSamples);
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this.bufferStartFrame = Math.floor(options.startTime * this.targetSampleRate);
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// Set to ensure that if the buffer start frame lands on a fractional sample, that the first timestamp still
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// comes out as exactly startTime
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this.timestampOffset = options.startTime - this.bufferStartFrame / this.targetSampleRate;
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}
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/**
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@@ -186,6 +177,7 @@ export class AudioResampler {
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// they see fit.
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this.sourceSampleRate = audioSample.sampleRate;
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this.sourceNumberOfChannels = audioSample.numberOfChannels;
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this.startTime = audioSample.timestamp;
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// Pre-allocate temporary buffer for source data
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this.tempSourceBuffer = new Float32Array(this.sourceSampleRate * this.sourceNumberOfChannels);
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@@ -193,6 +185,8 @@ export class AudioResampler {
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this.doChannelMixerSetup();
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}
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assert(this.startTime !== null);
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const requiredSamples = audioSample.numberOfFrames * audioSample.numberOfChannels;
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this.ensureTempBufferSize(requiredSamples);
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@@ -201,8 +195,8 @@ export class AudioResampler {
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const sourceView = new Float32Array(this.tempSourceBuffer.buffer, 0, sourceDataSize / 4);
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audioSample.copyTo(sourceView, { planeIndex: 0, format: 'f32' });
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const inputStartTime = audioSample.timestamp;
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const inputEndTime = Math.min(audioSample.timestamp + audioSample.duration, this.endTime);
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const inputStartTime = audioSample.timestamp - this.startTime;
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const inputEndTime = inputStartTime + audioSample.duration;
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// Compute which output frames are affected by this sample
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const outputStartFrame = Math.floor(inputStartTime * this.targetSampleRate);
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@@ -266,17 +260,18 @@ export class AudioResampler {
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return; // Nothing to finalize
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}
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assert(this.startTime !== null);
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const samplesWritten = (this.maxWrittenFrame + 1) * this.targetNumberOfChannels;
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const outputData = new Float32Array(samplesWritten);
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outputData.set(this.outputBuffer.subarray(0, samplesWritten));
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const timestampSeconds = this.bufferStartFrame / this.targetSampleRate;
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const audioSample = new AudioSample({
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format: 'f32',
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sampleRate: this.targetSampleRate,
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numberOfChannels: this.targetNumberOfChannels,
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timestamp: timestampSeconds + this.timestampOffset,
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timestamp: this.startTime + this.bufferStartFrame / this.targetSampleRate,
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data: outputData,
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});
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