From 165f97e4a06755cea8d212c32b7f48721316dac5 Mon Sep 17 00:00:00 2001 From: winlin Date: Fri, 30 Apr 2021 17:04:57 +0800 Subject: [PATCH 1/2] SquashSRS3: Link source flv in doc --- README.md | 4 ++-- trunk/doc/source.flv | 1 + 2 files changed, 3 insertions(+), 2 deletions(-) create mode 120000 trunk/doc/source.flv diff --git a/README.md b/README.md index 3b6b987f1..20f9c6fbf 100755 --- a/README.md +++ b/README.md @@ -25,10 +25,10 @@ docker run --rm -p 1935:1935 -p 1985:1985 -p 8080:8080 \ > To enable WebRTC, user MUST set the env `CANDIDATE`, see [#307](https://github.com/ossrs/srs/issues/307#issue-76908382). Open [http://localhost:8080/](http://localhost:8080/) to check it, then publish -[stream](https://github.com/ossrs/srs/blob/3.0release/trunk/doc/source.200kbps.768x320.flv) by: +[stream](https://github.com/ossrs/srs/blob/3.0release/trunk/doc/source.flv) by: ```bash -docker run --rm --network=host ossrs/srs:encoder ffmpeg -re -i ./doc/source.200kbps.768x320.flv \ +docker run --rm --network=host ossrs/srs:encoder ffmpeg -re -i ./doc/source.flv \ -c copy -f flv -y rtmp://localhost/live/livestream ``` > Note: If WebRTC enabled, you can publish by [H5](http://localhost:8080/players/rtc_publisher.html?autostart=true). diff --git a/trunk/doc/source.flv b/trunk/doc/source.flv new file mode 120000 index 000000000..32d9ecfc2 --- /dev/null +++ b/trunk/doc/source.flv @@ -0,0 +1 @@ +source.200kbps.768x320.flv \ No newline at end of file From f4f616d4e9cbffdaf3e125d065cd596d90ad569f Mon Sep 17 00:00:00 2001 From: winlin Date: Fri, 30 Apr 2021 17:22:39 +0800 Subject: [PATCH 2/2] Update README --- README.md | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/README.md b/README.md index 20f9c6fbf..b40276857 100755 --- a/README.md +++ b/README.md @@ -127,6 +127,7 @@ Other important wiki: - [x] [Experimental] Support mux RTP/RTCP/DTLS/SRTP on one port for WebRTC, [#307][bug #307]. - [x] [Experimental] Support client address changing for WebRTC, [#307][bug #307]. - [x] [Experimental] Support transcode RTMP/AAC to WebRTC/Opus, [#307][bug #307]. +- [x] [Experimental] Support AV1 codec for WebRTC, [#2324][bug #2324]. - [x] [Experimental] Enhance HTTP Stream Server for HTTP-FLV, HTTPS, HLS etc. [#1657][bug #1657]. - [x] [Experimental] Support push stream by GB28181, [#1500][bug #1500]. - [x] [Experimental] Support DVR in MP4 format, read [#738][bug #738]. @@ -157,7 +158,7 @@ Other important wiki: ## V4 changes -* v4.0, 2021-04-29, RTC: Support av1 for Chrome M90. 4.0.91 +* v4.0, 2021-04-29, RTC: Support AV1 for Chrome M90. 4.0.91 * v4.0, 2021-04-24, Change push-RTSP as deprecated feature. * v4.0, 2021-04-24, Player: Change the default from RTMP to HTTP-FLV. * v4.0, 2021-04-24, Disable CherryPy by --cherrypy=off. 4.0.90 @@ -1866,6 +1867,8 @@ Winlin [bug #1998]: https://github.com/ossrs/srs/issues/1998 [bug #2106]: https://github.com/ossrs/srs/issues/2106 [bug #2011]: https://github.com/ossrs/srs/issues/2011 +[bug #2324]: https://github.com/ossrs/srs/issues/2324 +[bug #1500]: https://github.com/ossrs/srs/issues/1500 [bug #zzzzzzzzzzzzz]: https://github.com/ossrs/srs/issues/zzzzzzzzzzzzz [exo #828]: https://github.com/google/ExoPlayer/pull/828