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Code Issues Releases Wiki Activity

Merge branch '4.0release' into merge/develop

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winlin 2021-06-30 07:29:22 +08:00
commit 2bd0deac47
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1
CHANGELOG.md
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@ -18,6 +18,7 @@ The changelog for SRS.
## SRS 4.0 Changelog
* v4.0, 2021-06-30, Merge [#2440](https://github.com/ossrs/srs/pull/2440), fix [#2390](https://github.com/ossrs/srs/issues/2390), SRT bug for zerolatency. 4.0.137
* v4.0, 2021-06-28, Merge [#2435](https://github.com/ossrs/srs/pull/2435), fix bug for HTTP-RAW-API to check vhost. 4.0.136
* v4.0, 2021-06-28, Fix [#2431](https://github.com/ossrs/srs/issues/2431), configure FFmpeg bug. 4.0.135
* v4.0, 2021-06-28, Merge [#2444](https://github.com/ossrs/srs/pull/2444), add libavcodec/crystalhd.c for FFmpeg. 4.0.134

36
README.md
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@ -106,26 +106,40 @@ A big THANK YOU goes to:
The ports used by SRS, kernel services:
* tcp://1935, for RTMP live streaming server([CN][v4_CN_DeliveryRTMP],[EN][v4_EN_DeliveryRTMP]).
* tcp://1985, HTTP API server, for HTTP-API([CN][v4_CN_HTTPApi], [EN][v4_EN_HTTPApi]), WebRTC([CN][v4_CN_WebRTC], [EN][v4_EN_WebRTC]), etc.
* tcp://8080, HTTP live streaming server, HTTP-FLV([CN][v4_CN_SampleHttpFlv], [EN][v4_EN_SampleHttpFlv]), HLS([CN][v4_CN_SampleHLS], [EN][v4_EN_SampleHLS]) as such.
* udp://8000, WebRTC Media([CN][v4_CN_WebRTC], [EN][v4_EN_WebRTC]) server.
* `tcp://1935`, for RTMP live streaming server([CN][v4_CN_DeliveryRTMP],[EN][v4_EN_DeliveryRTMP]).
* `tcp://1985`, HTTP API server, for HTTP-API([CN][v4_CN_HTTPApi], [EN][v4_EN_HTTPApi]), WebRTC([CN][v4_CN_WebRTC], [EN][v4_EN_WebRTC]), etc.
* `tcp://8080`, HTTP live streaming server, HTTP-FLV([CN][v4_CN_SampleHttpFlv], [EN][v4_EN_SampleHttpFlv]), HLS([CN][v4_CN_SampleHLS], [EN][v4_EN_SampleHLS]) as such.
* `udp://8000`, WebRTC Media([CN][v4_CN_WebRTC], [EN][v4_EN_WebRTC]) server.
For optional HTTPS services, which might be provided by other web servers:
* tcp://8088, HTTPS live streaming server.
* tcp://1990, HTTPS API server.
* `tcp://8088`, HTTPS live streaming server.
* `tcp://1990`, HTTPS API server.
For optional stream caster services, to push streams to SRS:
* udp://8935, Stream Caster: [Push MPEGTS over UDP](https://github.com/ossrs/srs/wiki/v4_CN_Streamer#push-mpeg-ts-over-udp) server.
* tcp://554, Stream Caster: [Push RTSP](https://github.com/ossrs/srs/wiki/v4_CN_Streamer#push-rtsp-to-srs) server.
* tcp://8936, Stream Caster: [Push HTTP-FLV](https://github.com/ossrs/srs/wiki/v4_CN_Streamer#push-http-flv-to-srs) server.
* udp://10080, Stream Caster: [Push SRT Media](https://github.com/ossrs/srs/issues/1147#issuecomment-577469119) server.
* `udp://8935`, Stream Caster: [Push MPEGTS over UDP](https://github.com/ossrs/srs/wiki/v4_CN_Streamer#push-mpeg-ts-over-udp) server.
* `tcp://554`, Stream Caster: [Push RTSP](https://github.com/ossrs/srs/wiki/v4_CN_Streamer#push-rtsp-to-srs) server.
* `tcp://8936`, Stream Caster: [Push HTTP-FLV](https://github.com/ossrs/srs/wiki/v4_CN_Streamer#push-http-flv-to-srs) server.
* `udp://10080`, Stream Caster: [Push SRT Media](https://github.com/ossrs/srs/issues/1147#issuecomment-577469119) server.
For external services to work with SRS:
* udp://1989, [WebRTC Signaling](https://github.com/ossrs/signaling#usage) server.
* `udp://1989`, [WebRTC Signaling](https://github.com/ossrs/signaling#usage) server.
## APIs
The API used by SRS:
* `/api/v1/` The HTTP API path.
* `/rtc/v1/` The HTTP API path for RTC.
* `/sig/v1/` The [demo signaling](https://github.com/ossrs/signaling) API.
Other API used by [ossrs.net](https://ossrs.net):
* `/gif/v1` The statistic API.
* `/service/v1/` The latest available version API.
* `/ws-service/v1/` The latest available version API, by websocket.
## Features

7
trunk/src/app/srs_app_latest_version.cpp
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@ -54,12 +54,7 @@ srs_error_t SrsLatestVersion::start()
return srs_success;
}
char buf[10];
srs_random_generate(buf, sizeof(buf));
for (int i = 0; i < (int)sizeof(buf); i++) {
buf[i] = 'a' + uint8_t(buf[i])%25;
}
server_id_ = string(buf, sizeof(buf));
server_id_ = srs_random_str(10);
return trd_->start();
}

12
trunk/src/app/srs_app_rtmp_conn.cpp
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@ -944,6 +944,11 @@ srs_error_t SrsRtmpConn::acquire_publish(SrsLiveSource* source)
SrsRequest* req = info->req;
// Check whether RTMP stream is busy.
if (!source->can_publish(info->edge)) {
return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "rtmp: stream %s is busy", req->get_stream_url().c_str());
}
// Check whether RTC stream is busy.
#ifdef SRS_RTC
SrsRtcSource *rtc = NULL;
@ -955,16 +960,11 @@ srs_error_t SrsRtmpConn::acquire_publish(SrsLiveSource* source)
}
if (!rtc->can_publish()) {
return srs_error_new(ERROR_RTC_SOURCE_BUSY, "rtc stream %s busy", req->get_stream_url().c_str());
return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "rtc stream %s busy", req->get_stream_url().c_str());
}
}
#endif
// Check whether RTMP stream is busy.
if (!source->can_publish(info->edge)) {
return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "rtmp: stream %s is busy", req->get_stream_url().c_str());
}
// Bridge to RTC streaming.
#if defined(SRS_RTC) && defined(SRS_FFMPEG_FIT)
if (rtc) {

2
trunk/src/core/srs_core_version4.hpp
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@ -9,6 +9,6 @@
#define VERSION_MAJOR 4
#define VERSION_MINOR 0
#define VERSION_REVISION 136
#define VERSION_REVISION 137
#endif