An API powered by Google's AI technology allows you to accurately convert speech into text. You can accurately caption your content, provide a better user experience with products using voice commands, and gain insight from customer interactions to improve your service. Google's deep learning neural network algorithms are the most advanced in automatic speech recognition (ASR). Speech-to-Text allows for experimentation, creation, management, and customization of custom resources. You can deploy speech recognition wherever you need it, whether it's in the cloud using the API or on-premises using Speech-to-Text O-Prem. You can customize speech recognition to translate domain-specific terms or rare words. Automated conversion of spoken numbers into addresses, years and currencies. Our user interface makes it easy to experiment with your speech audio.
Learn more
Any audio or video can be extracted to extract vocal, accompaniment, and other instruments. High-quality stem cutting based on the #1 AI-powered technology in the world. Next-generation vocal remover and music source separator service for fast, simple, and precise stem removal. You can remove vocal, instrumental, drums and bass tracks, as well as acoustic guitar, electric guitar, and synthesizer tracks, without any quality loss. You can start the service free of charge. Upgrade to get more files processed and faster results. Only for personal use. Move to the next level. You can process thousands of minutes of audio and/or video. This software is suitable for both personal and business use. Each LALAL.AI package has a limit on the amount of audio/video that can be split. The package minute limit is deducted from each file that has been fully split. You can split as many files you like, provided their total length does not exceed the minute limit.
Learn more
Gemini 3.8 Flash-Lite TTS
Gemini 3.8 Flash-Lite TTS is an expressive text-to-speech model from Google optimized for high-volume, cost-efficient speech generation. It is designed for workloads such as global dubbing, large-scale audio content production, localization, and conversational voice agents. Users can control characteristics such as tone, pacing, emphasis, and expressive nuance to shape how generated speech is delivered. Line-by-line direction allows scripts to include performance instructions and natural speech cues rather than producing uniformly spoken narration. The model supports long-form generation and is designed to preserve voice quality, natural pacing, and character consistency across extended audio. Native two-speaker staging allows developers and creators to generate multi-turn conversations while keeping speakers distinct and maintaining natural turn-taking. Scripted cues can introduce laughs, sighs, gasps, and listening responses such as “mhm” or “yeah” to make dialogue more conversational. Gemini 3.8 Flash-Lite TTS supports more than 100 languages and is designed for multilingual audio experiences at global scale, while generated Gemini Audio output includes SynthID watermarking for transparency. Developers can access the model through Google AI Studio and the Gemini API, with integration into Google Vids and planned API availability through Gemini Enterprise.
Learn more
Qwen3-TTS
Qwen3-TTS represents an innovative collection of advanced text-to-speech models created by the Qwen team at Alibaba Cloud, released under the Apache-2.0 license, which delivers stable, expressive, and real-time speech output with functionalities like voice cloning, voice design, and precise control over prosody and acoustic features. This suite supports ten prominent languages—Chinese, English, Japanese, Korean, German, French, Russian, Portuguese, Spanish, and Italian—along with various dialect-specific voice profiles, enabling adaptive management of tone, speech rate, and emotional delivery tailored to text semantics and user instructions. The architecture of Qwen3-TTS incorporates efficient tokenization and a dual-track design, facilitating ultra-low-latency streaming synthesis, with the first audio packet generated in approximately 97 milliseconds, making it ideal for interactive and real-time applications. Additionally, the range of models available offers diverse capabilities, such as rapid three-second voice cloning, customization of voice timbres, and voice design based on given instructions, ensuring versatility for users in many different scenarios. This flexibility in design and performance highlights the model's potential for a wide array of applications in both commercial and personal contexts.
Learn more
