webrtc-android

问题 I am trying to compile WebRTC Native Stack to build libwebrtc.aar but unfortunately unable to understand what's going wrong. System Information: Distributor ID: Ubuntu Description: Ubuntu 18.04.4 LTS Release: 18.04 Codename: bionic Java-Version: OpenJDK-8-JDK Python-Version: Python 2.7.17 I have followed the complete steps provided in the official WebRTC Native Development for Android. These steps have been completed without any interruption and every step took its time for completion. Then,

问题 I have implement an option of Video Conference on my application using the following example: https://github.com/androidthings/sample-videoRTC basically is is working very well but i have one major issue. the sreaming audio volume is very very low even when i put the maximum volume on my device. I have tried to check if there is any parameter that can define the audio volume but is was not able to find such parameters beside the AudioBitRate(=32) and the AudioCodec=("OPUS"). These are the

问题 I want to send unidirectional streaming data over a WebRTC datachannel, and is looking of the best configuration options (high BW, low latency/jitter) and others' experience with expected bitrates in this kind of application. My test program sends chunks of 2k, with a bufferedAmountLowThreshold event callback of 2k and calls send again until bufferedAmount exceeds 16k. Using this in Chrome, I achieve ~135Mbit/s on LAN and ~20Mbit/s from/to a remote connection, that has 100Mbit/s WAN