4.5 out of 5 stars
(446 Ratings)
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 about Google Cloud Speech-to-Text
4.5 out of 5 stars
(207 Ratings)
Log360 is a SIEM or security analytics solution that helps you combat threats on premises, in the cloud, or in a hybrid environment. It also helps organizations adhere to compliance mandates such as PCI DSS, HIPAA, GDPR and more. You can customize the solution to cater to your unique use cases and protect your sensitive data.
With Log360, you can monitor and audit activities that occur in your Active Directory, network devices, employee workstations, file servers, databases, Microsoft 365 environment, cloud services and more. Log360 correlates log data from different devices to detect complex attack patterns and advanced persistent threats. The solution also comes with a machine learning based behavioral analytics that detects user and entity behavior anomalies, and couples them with a risk score. The security analytics are presented in the form of more than 1000 pre-defined, actionable reports. Log forensics can be performed to get to the root cause of a security challenge.
The built-in incident management system allows you to automate the remediation response with intelligent workflows and integrations with popular ticketing tools.
Learn more about ManageEngine Log360
Gemini 2.5 Pro TTS
Gemini 2.5 Pro TTS represents Google's cutting-edge text-to-speech technology within the Gemini 2.5 series, designed to deliver high-quality and expressive speech synthesis tailored for structured audio generation needs. This model produces lifelike voice output that boasts improved expressiveness, tone modulation, pacing, and accurate pronunciation, allowing developers to specify style, accent, rhythm, and emotional subtleties through text prompts. Consequently, it is ideal for a variety of uses, including podcasts, audiobooks, customer support, educational tutorials, and multimedia storytelling that demand superior audio quality. Additionally, it accommodates both single and multiple speakers, facilitating varied voices and interactive dialogues within a single audio output, and supports speech synthesis in various languages while maintaining a consistent style. In contrast to faster alternatives like Flash TTS, the Pro TTS model focuses on delivering exceptional sound quality, rich expressiveness, and detailed control over voice characteristics. This emphasis on nuance and depth makes it a preferred choice for professionals seeking to enhance their audio content.
Learn more about Gemini 2.5 Pro TTS
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 about Qwen3-TTS
