| Ecosystem | 0 | 0 |
| Match Graph | 0 | 0 |
| Pricing | Free | Free |
| Capabilities | 5 decomposed | 12 decomposed |
| Times Matched | 0 | 0 |
Automatically extracts readable text content from web articles (via URL or direct paste) and converts it to audio using cloud-based text-to-speech synthesis. The system likely uses DOM parsing or content extraction libraries to isolate article body text while filtering navigation, ads, and metadata, then streams the extracted text to a TTS engine (possibly Google Cloud TTS, Azure Speech, or similar) for synthesis.
Unique: Combines automatic article extraction with TTS in a single freemium web interface, eliminating the manual copy-paste step required by generic TTS tools; appears to use intelligent content parsing to isolate article body rather than reading entire page HTML
vs alternatives: Faster workflow than browser TTS (no manual text selection) and more accessible than Natural Reader (freemium vs paid), but likely lower voice quality and no offline capability compared to premium competitors
Provides a voice selection interface allowing users to choose from multiple pre-synthesized voices (likely varying by gender, accent, age) and adjust playback parameters like speed and volume. This is implemented as a client-side audio player with voice selection mapped to different TTS voice IDs or pre-rendered audio variants, enabling real-time switching without re-synthesis.
Unique: Integrates voice selection and playback controls directly into the conversion interface rather than requiring separate audio player software; likely uses voice ID mapping to TTS provider's voice catalog (e.g., Google Cloud TTS voice names) for seamless switching
vs alternatives: More intuitive than command-line TTS tools or browser extensions requiring separate configuration; comparable to Pocket's voice feature but with explicit voice choice rather than single default voice
Implements a freemium model with usage limits (quota) for free users, likely tracking conversions per user via session cookies, local storage, or anonymous user IDs. The system enforces soft limits (e.g., 5 free conversions/month) before prompting upgrade, with a paid tier removing or significantly increasing limits. Backend likely uses a simple counter or rate-limiting middleware to track usage.
Unique: Removes barrier to entry with generous free tier (vs Natural Reader's limited trial), enabling casual users to test without credit card; quota tracking likely uses lightweight session-based approach rather than account-based metering
vs alternatives: More accessible than paid-only competitors (Natural Reader, Speechify) for initial testing; less restrictive than some freemium tools with 1-2 free conversions, but unclear if quota is competitive with browser TTS (which is free and unlimited)
Processes article-to-speech conversion with minimal latency, likely using a cloud TTS API (Google Cloud, Azure, or AWS Polly) with caching and streaming optimizations. The system probably queues synthesis requests, streams audio chunks to the client as they're generated, and caches frequently-converted articles to avoid re-synthesis. Architecture likely uses a serverless backend (Lambda, Cloud Functions) for cost-efficient scaling.
Unique: Optimizes for sub-10-second conversion time for typical articles by using cloud TTS APIs with streaming and caching, rather than local synthesis (which would be slower) or batch processing (which would delay playback)
vs alternatives: Faster than local TTS tools (e.g., espeak) due to cloud-based synthesis quality; comparable to Pocket's audio feature but with explicit freemium model and voice selection
Embeds an HTML5 audio player in the web interface with standard controls (play, pause, seek, volume) and likely persists playback position (current time, article ID) in browser local storage or session storage. This enables users to pause an article and resume from the same position on return, without requiring user accounts or backend state management.
Unique: Implements lightweight playback state persistence using browser local storage rather than requiring user accounts or backend state management, enabling frictionless resumption for casual users
vs alternatives: Simpler UX than Pocket (no account required for basic playback) but less feature-rich than dedicated audio apps (no cross-device sync, no history); comparable to browser TTS but with explicit player UI
Transcribes audio in 98 languages to text in the original language using a Transformer sequence-to-sequence architecture trained on 680,000 hours of diverse internet audio. The system uses mel spectrogram feature extraction via FFmpeg integration, processes audio through an AudioEncoder that generates embeddings, then applies an autoregressive TextDecoder with task-specific tokens to produce language-native transcriptions. Language-specific models (e.g., tiny.en, base.en) optimize for English-only workloads with reduced parameter count.
Unique: Unified multitasking Transformer model replaces traditional multi-stage speech pipelines (VAD → language detection → ASR → post-processing) with single forward pass; trained on 680K hours of internet audio providing robustness to background noise, accents, and technical speech unlike studio-trained competitors
vs alternatives: Outperforms Google Cloud Speech-to-Text and Azure Speech Services on non-English languages and noisy audio due to diverse training data; open-source allows local deployment without API latency or privacy concerns
Translates non-English speech directly to English text in a single forward pass using the same Transformer architecture as transcription, but with a translation task token prepended to the decoder input. The model learns to skip intermediate transcription and generate English output directly from audio embeddings, avoiding cascading errors from intermediate transcription steps. Supports 98 source languages translating to English only.
Unique: Direct audio-to-English translation without intermediate transcription step — the decoder learns to skip source language text generation and output English directly, reducing error propagation and latency compared to cascade approaches (transcribe → translate)
vs alternatives: Faster and more accurate than Google Translate + Google Speech-to-Text pipeline because it avoids intermediate transcription errors; open-source allows offline deployment unlike cloud translation APIs
Whisper Large v3 scores higher at 58/100 vs Article.Audio at 40/100.
Need something different?
Search the match graph →© 2026 Unfragile. Stronger through disorder.
Normalizes variable-length audio to exactly 30 seconds via `whisper.pad_or_trim()`: audio shorter than 30 seconds is padded with silence (zeros) to reach 30 seconds, audio longer than 30 seconds is trimmed to first 30 seconds. This ensures consistent input shape (80×3000 mel spectrogram) for the model, avoiding shape mismatches and enabling batch processing. Padding strategy is simple zero-padding rather than sophisticated techniques like repetition or interpolation.
Unique: Simple zero-padding strategy is computationally efficient and deterministic, but acoustically naive — alternative approaches (silence detection, repetition) not implemented in base library
vs alternatives: Simpler than librosa-based preprocessing with sophisticated padding; deterministic behavior aids reproducibility; zero-padding is fast but may introduce artifacts vs more sophisticated techniques
Returns transcription results as structured JSON objects containing: transcribed text, language code, duration, segments (with timing and text), and optional confidence metrics. The `model.transcribe()` API returns a dictionary with keys like 'text' (full transcript), 'language' (detected language), 'segments' (list of segment objects with start/end times and text). This structured format enables downstream processing (subtitle generation, database storage, API responses) without string parsing.
Unique: Structured output format is built into high-level API rather than requiring manual parsing — segments include timing and text, enabling direct use for subtitle generation or timeline-based applications
vs alternatives: More structured than raw text output; less detailed than forced alignment tools that provide phoneme-level information; JSON format is language-agnostic and integrates easily with web APIs
Detects the spoken language in audio by processing mel spectrograms through the AudioEncoder and using a language classification head that outputs probability distributions over 98 supported languages. The model leverages 680K hours of multilingual training data to recognize language characteristics from acoustic features alone, without requiring transcription. Language detection occurs as a preliminary step in the transcription pipeline and can be called independently via the language detection task token.
Unique: Language detection is integrated into the same Transformer model as transcription/translation via task tokens, allowing shared AudioEncoder computation and single model load — not a separate classifier, reducing memory footprint and inference overhead
vs alternatives: More accurate than acoustic-only language identification (e.g., librosa-based approaches) because it leverages semantic understanding from 680K hours of training; faster than transcription-based detection (identify language from first few words) because it uses acoustic features directly
Provides six model variants (tiny 39M, base 74M, small 244M, medium 769M, large 1550M, turbo 809M) with different parameter counts, VRAM requirements (1-10GB), and inference speeds (10x-1x relative to large). Each size trades accuracy for speed — tiny runs ~10x faster but with ~5-10% lower WER (word error rate), while large provides best accuracy at 10GB VRAM cost. Turbo variant (809M params) optimizes large-v3 for 8x speedup with minimal accuracy loss but lacks translation support.
Unique: Discrete model size family with published speed/accuracy/VRAM tradeoff matrix allows developers to make informed selection based on deployment constraints; turbo variant represents architectural optimization (knowledge distillation or pruning) achieving 8x speedup with <5% accuracy loss, distinct from simply using smaller base model
vs alternatives: More transparent tradeoff options than Whisper API (single model) or competitors like Deepgram (proprietary size selection); open-source allows local benchmarking on own hardware rather than relying on vendor performance claims
Automatically segments audio longer than 30 seconds into overlapping windows, processes each window independently through the transcription pipeline, and merges results with overlap handling to produce seamless full-length transcripts. The system uses `whisper.pad_or_trim()` to normalize each segment to exactly 30 seconds (padding with silence if needed), then applies the decoder to each segment and concatenates outputs while managing word-level boundaries and timestamp continuity across segment edges.
Unique: Sliding window approach with automatic overlap and boundary handling is built into high-level `model.transcribe()` API — developers don't manually implement segmentation, unlike lower-level APIs that require explicit window management
vs alternatives: Simpler than building custom segmentation logic; more robust than naive concatenation because it handles word-level boundary issues; faster than streaming approaches because it processes segments in parallel on GPU
Generates precise word-level timestamps (start and end times in milliseconds) for each word in the transcript by leveraging the decoder's attention weights and token alignment information. The system maps output tokens back to audio frames using the attention mechanism, then converts frame indices to millisecond timestamps based on the mel spectrogram hop length (20ms per frame). Timestamps are returned as part of the structured output alongside transcribed text.
Unique: Word-level timestamps are derived from attention weight alignment rather than separate timestamp prediction head — leverages existing decoder computation without additional model parameters, but introduces ±100-200ms uncertainty from frame quantization
vs alternatives: More granular than segment-level timestamps (which only mark 30-second boundaries); less accurate than forced alignment tools (e.g., Montreal Forced Aligner) but requires no phonetic lexicon or manual annotation
+4 more capabilities