Profile cover photo
Profile photo
WebRTC
12,036 followers -
Chrome WebRTC's official Google+ page.
Chrome WebRTC's official Google+ page.

12,036 followers
About
Communities and Collections
Posts

Post has attachment
Public
The WebRTC M68 release notes are now out: https://goo.gl/9aYUaZ. This release contains more than 20 new features, over 40 bug fixes and adds support for a new simulcast screen sharing mode. Additionally, developers can now test Unified Plan SDP in Chrome stable.
Add a comment...

Post has attachment
Public
WebRTC M66 release notes are now available. This release contains over 10 new features and over 40 bug fixes, enhancements and stability/performance improvements. Most important PSAs:

- a fix for long delays when Opus was used with DTX
- changes to RTCDTMFSender
- MediaStreamTrack.getCapabilities()
- RTCRtpTransceiver API

Detailed release notes can be found here: https://goo.gl/dMK8KE
Add a comment...

Post has attachment
Public
WebRTC M65 will be released in Chrome stable on March, 6. This release contains 10 new features and over 40 bug fixes. Important PSAs:

- support of RTCRtpSender.replaceTrack()
- Changed identifiers for some WebRTC stats
- Force HW H.264 to be Constrained Baseline Profile on Android
- Hard limit on the number of PeerConnections created in one process

Detailed release notes: https://goo.gl/svKXjr
Add a comment...

Post has attachment
Public
WebRTC M64 release notes are now out at https://goo.gl/YegTgN containing over 10 new features and 40 bug fixes.

Important PSA:

- significant parts of spec-compliant RTP Media API
- spec-compliant getUserMedia() error values
- video_frame_api split into two targets

Main new feature:

- improved bandwidth estimation for screen capture streams
Add a comment...

Post has attachment
WebRTC M62 release notes are published at https://goo.gl/D1R8o9. This release contains over 10 new features and 20 bug fixes.

Important PSA:
- prebuilt libraries for mobile development on Android and iOS are available at JCenter and cocoapods.org.
- PeerConnection::SetBitrate added to native PeerConnection API, limits the minimum and maximum bandwidth allocated for all RTP streams
Add a comment...

Post has attachment
WebRTC M61 release notes are finally out at https://goo.gl/8Afcg5. This release contains over 10 new features and over 40 bug fixes, enhancements and stability/performance improvements.

Important PSAs:
- getUserMedia from cross-origin iframes to be deprecated in Chrome M63
- webrtc/call.h removed from webrtc/
- Spec-compliant audio constraints for getUserMedia in Chrome M61

New important features:
- Mojo-based Video Capture Service
- <audio> elements now render audio without waiting for [optional] video frames

We encourage all developers to run versions of Chrome on the Canary, Dev, and Beta channels frequently and quickly report any issues found at bugs.webrtc.org
Add a comment...

Post has attachment
WebRTC M60 (!!!) release notes are now out at https://goo.gl/VYEHgM. The release contains over 10 new features and over 40 bug fixes, enhancements and stability/performance improvements.

Important fixes:
- Video jitter buffer fixes addressing freezes and decoding artifacts
- Issues with TCP + send-side bandwidth estimation fixed

Main features:
- Stopping audio track when audio input data is missing
- Updated VideoTrack labels for screen sharing
- Exposed new, standards-compliant getStats implementation in Java API
Add a comment...

Post has attachment
In 2017 we will make a significant push to finish WebRTC 1.0. This effort consists of:

1) Completing the WebRTC 1.0 specifications in both the W3C and IETF
2) Resolving remaining major spec incompatibilities in Chrome’s implementation
3) Ensuring interoperability across all browsers
4) Resolving remaining reliability pain points

Read more about our roadmap in this discuss-webrtc post:
Add a comment...

Post has attachment
WebRTC M59 release notes are published at https://goo.gl/BBfhTq. This release includes 15 new features and over 40 bug fixes. Most important new features are:

- Improved ICE candidate pooling
- New echo cancellation AEC3 available behind experimental flag
- AppRTCMobile can now be developed inside Android Studio
- Partial RTCRtpReceiver support with getContributingSources()
- More robust Bandwidth Estimation for cellular and WiFi connections.
- Spec-compliant video constraints processing in getUserMedia
- VP9 Denoiser enabled by default
Add a comment...

Post has attachment
WebRTC M58 release notes are published now at https://goo.gl/O8VW6m. This release contains over 20 new features and over 60 bug fixes, enhancements and stability/performance improvements. Main new features are:

- spec-compliant, promise-based RTCPeerConnection.getStats
- adding support of Opus 120ms encoding in WebRTC
- adding support for RTCPeerConnection.setConfiguration
- using WebRTC with authenticated HTTP proxy
- script for generating WebRTC Android Library (.aar)
- enable VP9 support in WebRTC HW decoders
- new video jitter buffer
- audio output debug recording
Add a comment...
Wait while more posts are being loaded