diff --git a/README.md b/README.md index 091bccd..40ebdaf 100644 --- a/README.md +++ b/README.md @@ -331,6 +331,26 @@ Steps for WASM debugging in Chrome: - Enable "WebAssembly Debugging: Enable DWARF support" in DevTools Experiments - DEBUG=1 make +## License + +This software is under the MIT open source license. + +[SPDX-License-Identifier: MIT](LICENSE) + +MicroPython for micro:bit is included in the build process via a submodule. + +We use dependencies via the NPM registry as specified by the package.json file under common Open Source licenses. + +Full details of each package can be found by running `license-checker`: + +```bash +$ npx license-checker --direct --summary --production +``` + +Omit the flags as desired to obtain more detail. + +A fork of libsamplerate_js to reduce bundle size is [hosted on GitHub](https://github.com/microbit-foundation/libsamplerate-js). + ## Code of Conduct Trust, partnership, simplicity and passion are our core values we live and diff --git a/package-lock.json b/package-lock.json index 8fab2c2..67484c9 100644 --- a/package-lock.json +++ b/package-lock.json @@ -8,6 +8,9 @@ "name": "@microbit-foundation/microbit-micropython-v2-simulator", "version": "0.1.0", "license": "MIT", + "dependencies": { + "@alexanderolsen/libsamplerate-js": "microbit-foundation/libsamplerate-js#v2.1.2-microbit.1" + }, "devDependencies": { "@types/emscripten": "^1.39.10", "esbuild": "^0.14.49", @@ -15,6 +18,43 @@ "vitest": "^0.22.1" } }, + "node_modules/@alexanderolsen/libsamplerate-js": { + "version": "2.1.2-microbit.1", + "resolved": "git+ssh://git@github.com/microbit-foundation/libsamplerate-js.git#14bdda3603133d21c07cfb48974617504576a7f7", + "license": "MIT" + }, + "node_modules/@esbuild/android-arm": { + "version": "0.15.18", + "resolved": "https://registry.npmjs.org/@esbuild/android-arm/-/android-arm-0.15.18.tgz", + "integrity": "sha512-5GT+kcs2WVGjVs7+boataCkO5Fg0y4kCjzkB5bAip7H4jfnOS3dA6KPiww9W1OEKTKeAcUVhdZGvgI65OXmUnw==", + "cpu": [ + 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"^0.14.49", "prettier": "2.6.0", "vitest": "^0.22.1" + }, + "dependencies": { + "@alexanderolsen/libsamplerate-js": "microbit-foundation/libsamplerate-js#v2.1.2-microbit.1" } } diff --git a/src/board/audio/index.ts b/src/board/audio/index.ts index 9dd127b..1fdb67e 100644 --- a/src/board/audio/index.ts +++ b/src/board/audio/index.ts @@ -1,6 +1,11 @@ +import { SRC } from "@alexanderolsen/libsamplerate-js/dist/src"; import { replaceBuiltinSound } from "./built-in-sounds"; import { SoundEmojiSynthesizer } from "./sound-emoji-synthesizer"; import { parseSoundEffects } from "./sound-expressions"; +import { + create as createSampleRateConverter, + ConverterType, +} from "@alexanderolsen/libsamplerate-js"; declare global { interface Window { @@ -11,7 +16,11 @@ declare global { interface AudioOptions { defaultAudioCallback: () => void; + defaultResampler: SRC; speechAudioCallback: () => void; + speechResampler: SRC; + soundExpressionResampler: SRC; + recordingResampler: SRC; } export class BoardAudio { @@ -24,6 +33,8 @@ export class BoardAudio { private muteNode: GainNode | undefined; private sensitivityNode: GainNode | undefined; + private recordingResampler: SRC | undefined; + default: BufferedAudio | undefined; speech: BufferedAudio | undefined; soundExpression: BufferedAudio | undefined; @@ -34,11 +45,17 @@ export class BoardAudio { initializeCallbacks({ defaultAudioCallback, + defaultResampler, speechAudioCallback, + speechResampler, + soundExpressionResampler, + recordingResampler, }: AudioOptions) { if (!this.context) { throw new Error("Context must be pre-created from a user event"); } + this.recordingResampler = recordingResampler; + this.muteNode = this.context.createGain(); this.muteNode.gain.setValueAtTime( this.muted ? 0 : 1, @@ -56,16 +73,19 @@ export class BoardAudio { this.default = new BufferedAudio( this.context, this.volumeNode, + defaultResampler, defaultAudioCallback ); this.speech = new BufferedAudio( this.context, this.volumeNode, + speechResampler, speechAudioCallback ); this.soundExpression = new BufferedAudio( this.context, this.volumeNode, + soundExpressionResampler, () => { if (this.currentSoundExpressionCallback) { this.currentSoundExpressionCallback(); @@ -75,12 +95,13 @@ export class BoardAudio { } async createAudioContextFromUserInteraction(): Promise { + // If we set a 44.1kHz rate then we fail to connect to user media on Mac as it selects 48000 + // So we leave it at the default hoping it's most likely to match user media... + // Until there's progress on this there doesn't seem a better way: + // https://bugzilla.mozilla.org/show_bug.cgi?id=1674892 this.context = - this.context ?? - new (window.AudioContext || window.webkitAudioContext)({ - // The highest rate is the sound expression synth. - sampleRate: 44100, - }); + this.context ?? new (window.AudioContext || window.webkitAudioContext)(); + if (this.context.state === "suspended") { return this.context.resume(); } @@ -92,21 +113,16 @@ export class BoardAudio { this.stopSoundExpression(); }; const synth = new SoundEmojiSynthesizer(0, onDone); + this.soundExpression!.setSampleRate(synth.sampleRate); synth.play(soundEffects); const callback = () => { const source = synth.pull(); if (this.context) { - // Use createBuffer instead of new AudioBuffer to support Safari 14.0. - const target = this.context.createBuffer( - 1, - source.length, - synth.sampleRate - ); - const channel = target.getChannelData(0); + const target = new Float32Array(source.length); for (let i = 0; i < source.length; i++) { // Buffer is (0, 1023) we need to map it to (-1, 1) - channel[i] = (source[i] - 512) / 512; + target[i] = (source[i] - 512) / 512; } this.soundExpression!.writeData(target); } @@ -197,6 +213,7 @@ export class BoardAudio { try { micStream = await navigator.mediaDevices.getUserMedia({ video: false, + // It seems Firefox ignores the rate set here audio: true, }); } catch (e) { @@ -208,41 +225,34 @@ export class BoardAudio { const source = this.context!.createMediaStreamSource(micStream); source.connect(this.sensitivityNode!); - // TODO: consider AudioWorklet - worth it? Browser support? - // consider alternative resampling approaches - // what sample rates are actually supported this way? + const recorder = this.context!.createScriptProcessor(2048, 1, 1); + + const inputSampleRate = this.context!.sampleRate; + this.recordingResampler!.inputSampleRate = inputSampleRate; + this.recordingResampler!.outputSampleRate = sampleRate; + recorder.onaudioprocess = (e) => { - const offlineContext = new (window.OfflineAudioContext || - window.webkitOfflineAudioContext)( - 1, - sampleRate * (e.inputBuffer.length / e.inputBuffer.sampleRate), - sampleRate + const resampled = this.recordingResampler!.full( + e.inputBuffer.getChannelData(0) ); - const source = offlineContext.createBufferSource(); - source.buffer = e.inputBuffer; - source.connect(offlineContext.destination); - source.start(); - offlineContext.addEventListener("complete", (e) => { - onChunk(e.renderedBuffer.getChannelData(0)); - samplesSent += e.renderedBuffer.length; - if (samplesSent >= samplesNeeded) { - this.stopRecording(); - } - }); - offlineContext.startRendering(); + onChunk(resampled); + samplesSent += resampled.length; + if (samplesSent >= samplesNeeded) { + this.stopRecording(); + } }; - this.sensitivityNode!.connect(recorder); - recorder.connect(this.context!.destination); - this.stopActiveRecording = () => { recorder.disconnect(); this.sensitivityNode!.disconnect(); source.disconnect(); - micStream.getTracks().forEach((track) => track.stop()); + micStream?.getTracks().forEach((track) => track.stop()); this.microphoneEl.style.display = "none"; this.stopActiveRecording = undefined; }; + + this.sensitivityNode!.connect(recorder); + recorder.connect(this.context!.destination); } boardStopped() { @@ -263,31 +273,35 @@ export class BoardAudio { class BufferedAudio { nextStartTime: number = -1; - private sampleRate: number = -1; constructor( private context: AudioContext, private destination: AudioNode, + private resampler: SRC, private callback: () => void - ) {} + ) { + this.resampler.outputSampleRate = this.context.sampleRate; + } init(sampleRate: number) { // This is called for each new audio source so don't reset nextStartTime // or we start to overlap audio - this.sampleRate = sampleRate; - } - - createBuffer(length: number) { - // Use createBuffer instead of new AudioBuffer to support Safari 14.0. - return this.context.createBuffer(1, length, this.sampleRate); + this.setSampleRate(sampleRate); } setSampleRate(sampleRate: number) { - this.sampleRate = sampleRate; + this.resampler.inputSampleRate = sampleRate; } - writeData(buffer: AudioBuffer) { - // Use createBufferSource instead of new AudioBufferSourceNode to support Safari 14.0. + writeData(data: Float32Array) { + // In practice the supported range is less than the 8k..96k required by the spec and varies by browser + // for a consistent performance profile we're always resampling for now rather than letting Web Audio do it + let sampleRate = this.context.sampleRate; + data = this.resampler.full(data); + + // Use createXXX instead to support Safari 14.0. + const buffer = this.context.createBuffer(1, data.length, sampleRate); + buffer.copyToChannel(data, 0); const source = this.context.createBufferSource(); source.buffer = buffer; source.onended = this.callCallback; diff --git a/src/board/conversions.ts b/src/board/conversions.ts index 392d780..f42bca7 100644 --- a/src/board/conversions.ts +++ b/src/board/conversions.ts @@ -98,12 +98,11 @@ export function convertAccelerometerNumberToString(value: number): string { export const convertAudioBuffer = ( heap: Uint8Array, source: number, - target: AudioBuffer + target: Float32Array ) => { - const channel = target.getChannelData(0); - for (let i = 0; i < channel.length; ++i) { + for (let i = 0; i < target.length; ++i) { // Convert from uint8 to -1..+1 float. - channel[i] = (heap[source + i] / 255) * 2 - 1; + target[i] = (heap[source + i] / 255) * 2 - 1; } return target; }; diff --git a/src/board/index.ts b/src/board/index.ts index fa9f543..e3752c2 100644 --- a/src/board/index.ts +++ b/src/board/index.ts @@ -1,3 +1,4 @@ +import { create as createResampler } from "@alexanderolsen/libsamplerate-js"; import svgText from "../microbit-drawing.svg"; import { Accelerometer } from "./accelerometer"; import { BoardAudio } from "./audio"; @@ -247,10 +248,22 @@ export class Board { noInitialRun: true, instantiateWasm, }); + + // We update the sample rates before use. + const recordingResampler = await createResampler(1, 48000, 48000); + const defaultResampler = await createResampler(1, 48000, 48000); + const speechResampler = await createResampler(1, 48000, 48000); + // Probably this one is never used so would be nice to avoid + const soundExpressionResampler = await createResampler(1, 48000, 48000); + const module = new ModuleWrapper(wrapped); this.audio.initializeCallbacks({ defaultAudioCallback: wrapped._microbit_hal_audio_raw_ready_callback, + defaultResampler, speechAudioCallback: wrapped._microbit_hal_audio_speech_ready_callback, + speechResampler, + soundExpressionResampler, + recordingResampler, }); this.accelerometer.initializeCallbacks( wrapped._microbit_hal_gesture_callback diff --git a/src/demo.html b/src/demo.html index a887cd8..8f3cd3f 100644 --- a/src/demo.html +++ b/src/demo.html @@ -99,6 +99,7 @@

MicroPython-micro:bit simulator example embedding

+ diff --git a/src/examples/record.py b/src/examples/record.py index 6c49d0f..521b244 100644 --- a/src/examples/record.py +++ b/src/examples/record.py @@ -4,7 +4,8 @@ rate_index = 0 print("Recording...") -my_track = microphone.record(3000) +my_recording = microphone.record(3000) +my_track = my_recording.track() print("Button A to play") while True: if button_a.was_pressed(): diff --git a/src/examples/record_background.py b/src/examples/record_background.py new file mode 100644 index 0000000..2d06859 --- /dev/null +++ b/src/examples/record_background.py @@ -0,0 +1,19 @@ +from microbit import microphone, audio, button_a, button_b, sleep + +rates = [7812, 3906, 15624] +rate_index = 0 + +print("Recording...") +my_recording = audio.AudioRecording(3000) +my_track = microphone.record_into(my_recording, wait=False) +sleep(3000) +print("Button A to play") +while True: + if button_a.was_pressed(): + audio.play(my_track, wait=False) + print("Rate playing", rates[rate_index]) + + if button_b.was_pressed(): + rate_index = (rate_index + 1) % len(rates) + print("Rate change to", rates[rate_index]) + my_track.set_rate(rates[rate_index]) \ No newline at end of file diff --git a/src/jshal.js b/src/jshal.js index c563973..160cc9c 100644 --- a/src/jshal.js +++ b/src/jshal.js @@ -230,8 +230,7 @@ mergeInto(LibraryManager.library, { Module.conversions.convertAudioBuffer( Module.HEAPU8, buf, - // @ts-expect-error - Module.board.audio.default.createBuffer(num_samples) + new Float32Array(num_samples) ) ); }, @@ -246,21 +245,10 @@ mergeInto(LibraryManager.library, { /** @type {number} */ num_samples ) { /** @type {AudioBuffer | undefined} */ let webAudioBuffer; - try { - // @ts-expect-error - webAudioBuffer = Module.board.audio.speech.createBuffer(num_samples); - } catch (e) { - // Swallow error on older Safari to keep the sim in a good state. - // @ts-expect-error - if (e.name === "NotSupportedError") { - return; - } else { - throw e; - } - } + const jsBuf = new Float32Array(num_samples); // @ts-expect-error Module.board.audio.speech.writeData( - Module.conversions.convertAudioBuffer(Module.HEAPU8, buf, webAudioBuffer) + Module.conversions.convertAudioBuffer(Module.HEAPU8, buf, jsBuf) ); }, diff --git a/src/mpconfigport.h b/src/mpconfigport.h index 3ae7f20..aa73d9c 100644 --- a/src/mpconfigport.h +++ b/src/mpconfigport.h @@ -129,6 +129,7 @@ extern uint32_t rng_generate_random_word(void); ((mp_raise_NotImplementedError(MP_ERROR_TEXT("simulator limitation: asm_thumb code"))), p) // The latency of fetching 32 byte audio frames is too much so increase the size -#define AUDIO_OUTPUT_BUFFER_SIZE (64) +// 128 matches the web audio internal buffer sizes +#define AUDIO_OUTPUT_BUFFER_SIZE (128) #endif