WebRTC M76目前包含3个新功能和50多个错误修复,增强功能和稳定性/性能改进。与之前的版本一样,官方鼓励所有开发人员经常在Canary,Dev和Beta渠道上运行Chrome版本,并快速报告发现的任何问题。
特征
实现了RTCRtpTransceiver.setCodecPreferences()
此新功能允许开发人员选择协商呼叫的编解码器,可能会删除默认编解码器或更改其首选顺序。这也可用于禁用RTX,RED或FEC编解码器条目。
实现了RTCSctpTransport
这允许检查用于DataChannel的传输状态。
实现了RTCRtpSender.setStreams
这允许设置与发送者的轨道相关联的MediaStream。
弃用
问题 |
描述 |
零件 |
弃用UMA指标WebRTC.Audio。{AecDelayAdjustmentMsSystemValue,AecDelayAdjustmentMsAgnosticValue} |
音频 |
|
删除StringRtpHeaderExtension |
网络> RTP |
|
消除WebRtcRTPHeader的使用 |
网络> RTP |
功能和Bug修改
Type |
Issue |
Description |
Component |
Feature |
Add rtpTimestamp to contributing sources |
Network>RTP |
|
Feature |
Add FrameMarking RTP Header Extension support in H.264 receiver |
Video |
|
Feature |
Add the possibility to limit the delay based bandwidth estimator to increase |
Network |
|
Feature |
Create superframe index header and append it to frame buffer |
Video |
|
Feature |
[standard stats] Implement stats for roundTripTime of RTP streams of kind video |
Stats |
|
Feature |
[standard stats] Implement stats for roundTripTime of RTP streams of kind audio |
Stats |
|
Feature |
[standard stats] Implement stats for resolution and framerate pre-encoding |
Stats |
|
Feature |
[standard stats] Implement stats for quality limitation: qualityLimitationReason |
Stats |
|
Feature |
[standard stats] Implement jitterBufferDelay and jitterBufferEmittedCount for video |
Stats |
|
Feature |
[standard stats] Implement counters for retransmitted bytes |
Stats |
|
Feature |
[standard stats] Implement stats for target encode bitrate |
Stats |
|
Feature |
[standard stats] Implement stats for error correction of RTP streams |
Stats |
|
Feature |
[standard stats] Implement stats for audible/silent concealed samples |
Stats |
|
Feature |
[standard stats] Implement stats for accelerating/decelerating playout speed |
Stats |
|
Feature |
Makes send packet information non optional for feedback reports. |
BWE |
|
Feature |
Split voe::Channel into send and receive classes for audio rtp transport. |
Audio, Network>RTP |
|
Feature |
Implement RTCRtpTransceiver::setCodecPreferences() |
PeerConnection |
|
Feature |
Implement most of RTCRemoteInboundRtpStreamStats |
Stats |
|
Feature |
SCTP SDP m-lines: Convert to sending new draft SDP spec |
PeerConnection |
|
Feature |
[standard stats] Implement stats for packet send-side delay |
Stats |
|
Feature |
Add RTP timestamp to RTCRtpReceiver::RTCRtpContributingSource |
Blink>WebRTC>Network |
|
Feature |
[Video Capture, Feature] Dynamic Screen Capture |
Blink>WebRTC>Video |
|
Feature |
Implement RTCRtpTransceiver.setCodecPreferences() |
Blink>WebRTC>PeerConnection |
|
Feature |
mDNS service for IP handling in WebRTC |
Blink>WebRTC>Network |
|
Feature |
Implement RTCRtpSender.setStreams() |
Blink>WebRTC>PeerConnection |
|
Feature |
Implement RTCSctpTransport |
Blink>WebRTC>PeerConnection |
|
Bug |
packetization mode should be checked when selecting H264 as send video codec |
PeerConnection |
|
Bug |
VP8 Decoder: Quality expectation and improvements for Accelerated Decoders in chromium |
Blink>WebRTC>Video |
|
Bug |
Make sure packets in the pacer queue are preserved |
Network>RTP |
|
Bug |
Potential overflow in sequence number map tracking loss vectors |
Network>RTP |
|
Bug |
Merge to M75: Write VP9 RTP SS on key frames of each independently coded spatial layer. |
Blink>WebRTC>Video |
|
Bug |
Potentially unnecessary scaling in LibvpxVp8Encoder::Encode() |
||
Bug |
In sumulcast mode VP9 sender doesn't write scalability structure on key frames of high spatial layers |
Video |
|
Bug |
Duplicate calls to OnSentPacket() breaks ALR detection |
BWE |
|
Bug |
Incoming offer for simulcast does not generate video |
PeerConnection |
|
Bug |
The runtime-settings in aecdumps for the pre-amplifier gain cannot be overruled in audioproc_f |
Audio |
|
Bug |
Color space not parsed correctly on receiver side |
Network>RTP |
|
Bug |
Acknowledged bitrate estimate can get stuck at low bandwidth. |
||
Bug |
Populate meta-information fields on VideoFrame from EncodedImage in one place |
Video |
|
Bug |
Surface max number of channels in SctpTransport interface |
Blink>WebRTC>PeerConnection |
|
Bug |
Surface max message size in RTCSctpTransport |
Blink>WebRTC>PeerConnection |
|
Bug |
Surface remote certificates in RtcDtlsTransport |
Blink>WebRTC>PeerConnection |
|
Bug |
CriticalSection doesn't play well with audio callback threads on MacOS |
Audio, Internals |
|
Bug |
RTT based backoff is not capped below. |
||
Bug |
SCTP: Compute max message size and max channels correctly |
DataChannel |
|
Bug |
Clock implementation hides mutable behavior hidden under const. |
||
Bug |
Bandwidth toggles between two estimates in StartUpPhase. |
BWE |
|
Bug |
The minimum comfort noise level in AEC3 is too high |
Audio |
|
Bug |
Excessive AEC suppression |
Audio |
|
Bug |
PeerConnectionInterface doesn't expose any useful error information |
PeerConnection |
|
Bug |
Need a way to add unstandardized stats for native applications |
Stats |
|
Bug |
rfc6184, rfc6185 sprop-parameter-sets |
Video |
|
Bug |
Simulcast streams will send one key frame on all spatial layers for each FIR with different SSRC |
Video |
|
Bug |
SDP parsing: addIceCandidate with candidate priority 0 is not rejected |
Network |
|
Bug |
Make unpack_aecdump unpack RuntimeSettings |
Audio |
|
Bug |
Let RuntimeSetting store either int or float |
Audio |
|
Bug |
Add RuntimeSetting for volume change |
Audio |
|
Bug |
MediaPicker: don't use TabbedPane when there's only one tab |
Blink>GetUserMedia>Desktop |
|
Bug |
Webcam.js Error: Webcam is not loaded yet |
Blink>GetUserMedia>Webcam |
|
Bug |
Unable to access microphone on Huawei Matebook X Pro |
Blink>GetUserMedia>Mic |
|
Bug |
MediaDeviceInfo object of kind videoinput is missing groupId |
Blink>GetUserMedia |
|
Bug |
Merge to M75: VP9 low-fps screen share fixes |
Blink>WebRTC>Video |
|
Bug |
addIceCandidate(new RTCIceCandidate({candidate, sdpMid})) no longer works |
Blink>WebRTC>PeerConnection |
|
Bug |
Distorted sound when using Web Audio API to mux audio sources in WebRTC on Mac |
Blink>WebRTC |
|
Bug |
Chrome uses obsolete format for SCTP data channels |
Blink>WebRTC>PeerConnection |
|
Bug |
Merge Request for [ Ensure that we always set values for min and max audio bitrate.] |
Blink>WebRTC>Audio |
|
Bug |
Unreasonable IO buffer size on Mac audio output when unplugging device |
Blink>Media>Audio, Blink>WebRTC>Audio, Internals>Media>Audio |
|
Bug |
Tab mirroring audio quality is significantly worse with audio service enabled on M75, 76 |
Blink>WebRTC>Audio, Internals>Cast>Streaming, Internals>Media>Audio, Internals>Media>Capture |
|
Bug |
iceConnectionState not going past "checking" in M75 |
Blink>WebRTC |
|
Bug |
Merge to M75: Expand UsagePattern and private IP address definition |
Blink>WebRTC>Network |
|
Bug |
Merge to M75: Parse color space only in last packet of key frame |
Blink>WebRTC>Video |
|
Bug |
Remove generic error from WebRTC event log collection |
Blink>WebRTC>Tools |
|
Bug |
Invoking getStats with an invalid second argument (such as errorCallback) is no longer equivalent to getStats(successCallback) |
Blink>WebRTC |
|
Bug |
WebRtcRemoteEventLogManager does not always upload over WiFi |
Blink>WebRTC>Tools |
|
Bug |
RTCPeerConnection.onnegotiationneeded can sometimes fire multiple times in a row |
Blink>WebRTC>PeerConnection |
|
Bug |
Microphone doesn't work |
Blink>GetUserMedia>Mic, OS>Kernel>Camera, Platform>Apps>Hangouts |
|
Bug |
Video feed from Brio 4K camera is flickering on Jaq device |
Blink>GetUserMedia>Webcam, OS>Kernel>Camera |