WebRTC[26]-WebRTC M76发行说明
目录
实现了RTCRtpTransceiver.setCodecPreferences（）
前言
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 sequen |
剩余60%内容，订阅专栏后可继续查看/也可单篇购买
WebRTC 作为当下最热门的实时音视频通讯框架，涉及非常多的过程，比如采集、编码、组包、发包、传输、收包、丢包重传、解封装、解码、音视频同步、渲染等，同时还包括很多功能特性，比如ANS、AGC、AEC，REMB、GCC、CNG、FEC、PLI、SVC等，需要一点点深入理解其中的奥秘。
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