目录

前言

重要特性

修复内容

前言

WebRTC M75同样提供了Android和iOS的版本库，它包含3个新功能和50多个错误修复，增强了相关的功能和稳定性，以及一些性能的改进措施。

《WebRTC工作原理精讲》系列-总览

重要特性

1. RTPsenders和RTPReceiver支持了IEC和DTLS的设置接口。

2. chrome://webrtc-internals包含了标准的getStats()信息以及其他一些新的状态信息。

3. 完善了PeerConnection的negotiationneeded事件通知机制，之前的版本通知应用程序需要进行会话协商存在一定问题。

修复内容

Issue

Description

Component

9688

Simulcast streams will send one key frame on all spatial layers for each FIR with different SSRC

Video

9934

Makes send packet information non optional for feedback reports.

BWE

10081

Unify congestion window and pacing buffer pushbacks.

BWE

10171

Include pacing buffer size in congestion window pushback.

BWE

10259

Create FrameBufferController interface and allow its injection for VP8

Video

10366

Make AEC3 the default AEC option in WebRTC

Audio

8420

H264 constrained baseline fails to be decoded

Video

8434

Excessive AEC suppression

Audio

9033

Pass coded_size info along in webrtc::VideoFrame

Video

10222

Bandwidth toggles between two estimates in StartUpPhase.

BWE

10325

PacedSender send to much padding when there are small packets sent

BWE

10341

AEC3: Echo during onsets

Audio

10368

RTT based backoff is not capped below.

BWE

10383

Simulcast video sends SDES with CNAME items with zero length

Network>RTP, PeerConnection

10392

Postpone decoding after expand causes too much delay in high packet loss scenarios

Audio

10393

Add support for writing a call order file in audioproc_f

Audio

10409

Fuzzing for simulcast

PeerConnection

10413

Add histograms to bandwidth probing code

BWE

10415

Increase default maximum jitter buffer size

Audio

10427

Make keyframe generation/request intervals tuneable

Video

10436

The way OpenSLEngineManager is shared between OpenSLESPlayer and OpenSLESRecorder is unsafe

Audio

10437

Adopt INTER_LAYER_PRED_OFF

Video

10439

Provide common interface for bitstream parsers

Video

10447

[standard stats] Implement counters for retransmitted bytes

Stats

10448

[standard stats] Implement totalEncodeTime

Stats

10449

[standard stats] Implement lastPacketReceivedTimestamp

Stats

10452

[standard stats] Implement stat for content type

Stats

10462

Acknowledged bitrate estimate can get stuck at low bandwidth.

BWE

10463

AEC3: missing bound checks when accessing a vector in the signal dependent erle estimator code

Audio

10475

RTCP XR target bitrate could be incorrect.

Video

10491

addTransceiver doesn't validate input rids

PeerConnection

10493

Fix timeouts in replay fuzzers.

Blink>WebRTC>Network

10501

Pass information from incoming LossNotification RTCP messages to video encoder

Video

10502

Duration of video pause is not included into sum of squared frame durations

Video

10533

The minimum comfort noise level in AEC3 is too high

Audio

10543

Color space not parsed correctly on receiver side

Network>RTP

10546

The runtime-settings in aecdumps for the pre-amplifier gain cannot be overruled in audioproc_f

Audio