Case Study: Improved Quality of Video and Audio Conferencing
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Improved quality of video and audio conferencing

The solutions we developed ensure high picture and sound quality during conferences and meetings held on a cloud collaboration platform.

Improved Quality of Video and Audio Conferencing

    Client and challenge

    Our client is an enterprise cloud communications and collaboration platform provider. Their platform offers a unified experience that seamlessly connects voice, video, messaging, and content sharing.

    The team at EffectiveSoft worked as an integral part of the client’s in-house team. They had to ensure high picture quality on low bitrates with minimal latency during one-on-one and group video calls. This was necessary because the quality of video and audio on such calls was previously impaired.

    The factors that contributed to the poor quality of video and audio included:

    • Poor Internet connection (low network bandwidth).
    • Diversity of devices used by attendees with different processing power and thus rendering capacity.
    • The large size of a high-definition video stream.
    • The old version of the WebRTC Voice Engine.
    • Client

      Cloud communications and collaboration platform provider
    • Solution

      Video and audio conferencing solution
    • Service

    • Domain

    • Technology

      WebRTC, C++, Objective-C
    • Outsourcing model


    Hardware video acceleration

    • Our team developed multiple solutions to improve the performance of multi-user video and audio conferencing. One of the methods we used to optimize the usage of device processing resources is hardware acceleration. This involves delegating encoding/decoding functions to the Graphics Processing Unit (GPU) instead of the Central Processing Unit (CPU).
    • We implemented a hardware encoder that supports devices on MS Windows and Mac OS with GPUs manufactured by Nvidia, Intel, and AMD. The team used native algorithms from GPU manufacturers and mobile devices for each particular case to ensure compatibility and efficiency.
    • To ensure high-quality audio streaming, we updated the platform to the latest WebRTC version. This migration has improved echo cancellation, noise reduction, and automatic microphone sensitivity adjustment, resulting in a better audio experience for users.
    • Our engineers have also integrated client-side audio mixing for one-on-one conferencing to reduce the load on the server and improve the performance of the platform. This feature allows users to mix their audio locally, reducing the amount of data that needs to be sent to the server, resulting in faster and more efficient conferencing.


    Tech stack

    • Languages:

    • Technologies:

      • Intel Quick Sync Video
      • WebRTC
      • DirectX
      • NVEnc
      • VideoToolbox
      • Android NDK
      • Advanced Media Framework
    • Tools:

      • MS Visual C++
      • CMake
      • Android Studio
      • XCode

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