标签：#Audio

Outstanding call quality and an excellent price point put these earbuds way ahead of pricier competition.

📝 Originally published (in Japanese) at forge.workstyle.tech . When building voice-based AI interactions in the browser (avatars, voice bots, streaming AI), you’ll inevitably hit pitfalls stemming from audio physics and browser implementation quirks. This article compiles 16 traps I encountered during product development , organized in a symptom → cause → solution lookup format . No need to read from top to bottom—jump straight to the symptom you’re facing. Echo and Self-Response Issues 1. Avatar Responds to Its Own Voice (Despite echoCancellation: true ) Symptom : TTS audio is picked up by the mic, and STT recognizes it as user speech, creating a self-response loop. Cause : AEC (Acoustic Echo Cancellation) requires a reference signal (the "sound to cancel"). Only the browser's official playback paths ( <audio> / WebRTC receiver tracks) serve as references. Custom playback via Web Audio API does not reliably function as a reference . Solution : Return TTS audio from the server as a WebRTC remote track and play it via an <audio> element. This eliminates echoes without text-matching workarounds (tested: 99 seconds of continuous speech with speakers on, zero false user turn detections). 2. Echoes Are Gone, but Speaking Simultaneously with the Avatar Distorts My Voice and Causes Misrecognition Symptom : Only during dual speech, proper nouns get mangled (e.g., "社員数" → "シャインズ"), especially at word beginnings. Cause : Fundamental AEC trade-off. To cancel echoes, AEC suppresses/distorts near-end (user) audio during dual speech. Solution : Mitigate in three layers: ① Increase mic Opus bitrate and enable FEC (see Pitfall 13) ② Provide vocabulary hints to STT (see separate article: use "recent avatar speech" as initial_prompt , not a dictionary) ③ Instruct LLM: "Input is STT transcription with potential errors. Interpret unnatural words as phonetically similar terms and add confirmation prompts." 3. Can’t Suppress Audio from Other Apps (Music, Videos) Symptom : Audio/lyrics fr

Two audio players do not make a fair before-and-after test. If the second player restarts from zero or takes half a second to load, the user is no longer comparing two versions of the same moment. They are comparing two memories. That is a weak way to evaluate any audio effect. It is especially weak for AI processing. A denoiser can remove a fan while softening consonants. A de-reverb model can reduce the room tail while making the voice sound less natural. The output may be cleaner without being better. The preview therefore has one job: let the listener switch quickly enough to hear both the improvement and the damage. The rule I use is deliberately boring. Both versions should contain the same edit and play from the same position. Switching should not restart playback or create a pause. The interface should not hint that one version is supposed to win. Two independent <audio> elements fail surprisingly quickly. Each owns its playback state, buffering behavior, clock, and seek operation. The user ends up finding the same position twice and comparing one sound with a memory of another. A better interface has one transport and one version control: [ Play ] [ Original | Processed ] 00:18 ━━━━━━━ 00:42 The transport decides where playback happens. The segmented control decides which signal is audible. One transport, two signals For a short preview, I decode both files into AudioBuffer s, start them at the same AudioContext time and offset, and route each through its own GainNode . Both sources run; only one gain is open. decodeAudioData() decodes complete file data and resamples it to the context's sample rate. The decoded buffers can then share the same audio clock. See the MDN documentation for format and loading details. The core is small: const context = new AudioContext (); const originalGain = context . createGain (); const processedGain = context . createGain (); originalGain . connect ( context . destination ); processedGain . connect ( context . destination )

Sennheiser announced the next version of its flagship wireless earbuds with a major redesign focusing on sustainability. The company claims the new design improves the Momentum True Wireless 5's "comfort, stability, and portability," but a more compelling reason to upgrade is that you won't have to replace them in a few years when they stop […]

We may have our first glimpse of Apple's rumored camera-equipped AirPods, thanks to a video that MacRumors found in the macOS Tahoe 26.7 Release Candidate. The short video clip features a man - who is wearing the new AirPods - holding up a book with the cover displayed, so that Visual Intelligence can see the […]

Sonos released an update to its mobile app that finally introduces support for iOS' Live Activities, giving iPhone users quick access to playback controls on their lockscreen. The added functionality is a "much requested, anticipated, and needed feature," according to a post Sonos shared to Reddit today that was spotted by 9to5Mac. While some features […]

Strings of code in Samsung's Galaxy Wearable app hint at an upcoming pair of over-ear headphones that could compete with the AirPods Max, SamMobile reports. Samsung's reportedly referring to the headphones as the "Galaxy H1," and SamMobile says they could launch sometime in 2027. That would make these the company's first pair of over-ear headphones […]

Today, I’m talking with Lila Snyder, who is the CEO of Bose. You certainly know Bose — it’s one of the most famous brands in all of consumer tech. The company started 60 years ago selling speakers to consumers, and its focus on research and development has led it to be a leader in both […]

Not every great pair of headphones belongs in the gym. These do, thanks to their secure fit, durable design, and impeccable sound.

Following the release of its premium and very expensive $650 The Collexion headphones in May, Sony could be taking its 1000X line of wireless noise-canceling headphones in a cheaper direction next month. The company is reportedly planning a refreshed and more affordable version of its six-year-old WH-1000XM4 headphones that will feature similar performance and specs […]

If you're building — or integrating — a browser-based audio trimmer, the question that eventually reaches your inbox isn't "does it cut audio?" The real question is: does it cut audio correctly across the inputs we actually receive from users? That shift, from feature presence to behavior under fuzzy conditions, is what turns a demo into a product. This article walks through the QA matrix I use when reviewing client-side trimmers before release, with an emphasis on the silent failures that don't show up in a happy-path recording. The tool under review for most of this article is the Lizely audio cutter ( in-depth walkthrough ), but the principles apply to any browser trimmer that decodes via AudioContext or OfflineAudioContext . What "Trim" Actually Means Once You Leave the Lab In the lab, you upload a 44.1 kHz stereo WAV, drag two handles, click export, and verify the output. In production, users upload M4A recordings from iPhone Voice Memos, AMR files from old Android handsets, mono 8 kHz captures from cheap conference mics, and — occasionally — files renamed from .wav to .mp3 without re-encoding. Each of those paths stresses a different layer of the pipeline. The first thing to test, before any UI work, is the decode step. Browsers expose this through the decodeAudioData method on BaseAudioContext , documented on MDN's BaseAudioContext page . MDN is explicit about something engineers often miss: decodeAudioData detaches the input ArrayBuffer . If your trimmer holds a reference to the original buffer for "undo" and reuses it, you'll decode an empty buffer the second time around and get a silent result. That's a real defect class, not a theoretical one. The second thing to test is what happens when decoding fails. The spec says decodeAudioData invokes the error callback with a DOMException , but the browser-specific error messages vary. Chrome tends to surface "Decoding error" with no detail; Firefox appends the underlying codec name. Your QA suite should assert on

Bose announced a second-generation version of its entry-level QuietComfort noise-canceling headphones that were originally introduced in late 2023. The new headphones feature a refreshed design with a more comfortable headband that's easier to adjust and ANC that better adapts to changing noise environments. The second-gen QuietComfort are also getting Bose's TrueSpatial immersive audio technology that […]

Open earbuds are having a bit of a moment right now, and Nothing is the latest company jumping on the trend. Its budget sub-brand has introduced the CMF Clip Pro, CMF's first open-ear buds that are designed to provide comfort and sound quality for people who don't want to sacrifice their situational awareness. The $99 […]

Whether you want an audio player that’ll grow with your kid or one that your toddler can operate independently, I have a recommendation for you.

The $100 earbuds look and sound better than earbuds nearly twice the price—as long as you can stomach just average noise canceling.

moin, ich möchte euch mein aktuelles Projekt vorstellen: Die Wiederbelebung eines Alienware m17x (Baujahr 2008) als dedizierte DJ-Workstation unter Linux. Ziel war es, alte Hardware nachhaltig zu nutzen und eine stabile Umgebung für Mixxx zu schaffen. Die Hardware: Notebook: Alienware m17x (Core 2 Duo, 4GB RAM, SSD) OS: KDE Neon mit Low-Latency-Kernel (6.8.0) Software: Mixxx 2.4 Audio-Interface: Günstiges USB-Audio-Device für den Master-Ausgang Das Problem: Mixxx verweigerte unter ALSA den Dienst mit der Fehlermeldung: Error opening "USB Audio Device (hw:1,0)" - Invalid sample rate Die Analyse über /proc/asound/card1/stream0 zeigte die Ursache: Das USB-Gerät unterstützt ausschließlich 46875 Hz – eine für Audio-Interfaces sehr unübliche Rate, die weder 44100 Hz noch 48000 Hz entspricht. Der direkte Zugriff über hw:CARD=Device,DEV=0 schlug fehl. Die Lösung: Die Rettung war die Aktivierung der ALSA-Plug-Erweiterung über PipeWire/ALSA, die eine automatische Sample-Rate-Konvertierung erlaubt. Starten Sie Mixxx nicht direkt, sondern setzen Sie zuvor die Umgebungsvariable: export PA_ALSA_PLUGHW=1 mixxx Damit Mixxx auch dauerhaft korrekt startet (z.B. über das KDE-Menü), habe ich den Starter wie folgt angepasst: bash -c "export PA_ALSA_PLUGHW=1; mixxx" Ergebnis: ✅ Master-Ausgabe über das USB-Device funktioniert stabil. ✅ Kopfhörer-Vorhören (C-Media USB Headphone Set) läuft parallel. ✅ Das System läuft trotz des Alters der Hardware (2008) flüssig und mit geringer Latenz. Die vollständige Dokumentation inklusive Fotos des Umbaus, der genauen Kernel-Einstellungen und der Konfiguration findet ihr in meinem Open-Source-Repository: 👉 [ https://github.com/qrishii/DJ-Installationen ] Ich hoffe, diese Lösung hilft anderen, die ähnliche Probleme mit exotischen USB-Audio-Raten unter Linux haben! das Leben ist lustig

Its latest Pro headset can mix between four inputs and has improved wireless connectivity over the originals.

The company’s first headphones look very stylish and have great tactile features, but underwhelm on sound quality.

Tech companies love to sell us expensive gadgets to solve all of life's little problems. Sleepbuds sold by the likes of Anker and Ozlo are a good example. These miniature marvels of engineering sit flush in the ear, and allow side-sleepers to doze off listening to podcasts, audiobooks, music, or white noise without annoying their […]

The Media Router is a tool that automatically selects the best image, video, or audio generation model for a request based on whether a developer prioritizes quality, speed or cost.