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RTC: Refine config, aac to rtmp_to_rtc, bframe to keep_bframe. v4.0.174
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parent
fe9e43b6d4
commit
71ed6e5dc5
16 changed files with 94 additions and 52 deletions
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@ -33,7 +33,7 @@ vhost __defaultVhost__ {
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rtc {
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enabled on;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
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aac transcode;
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rtmp_to_rtc on;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
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rtc_to_rtmp on;
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}
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@ -31,7 +31,7 @@ vhost __defaultVhost__ {
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rtc {
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enabled on;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
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aac discard;
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rtmp_to_rtc off;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
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rtc_to_rtmp off;
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}
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@ -459,19 +459,16 @@ vhost rtc.vhost.srs.com {
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# default: 0
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drop_for_pt 0;
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###############################################################
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# For transmuxing RTMP to RTC, the strategy for bframe.
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# keep Keep bframe, which may make browser with playing problems.
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# discard Discard bframe, maybe cause browser with little problems.
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# default: discard
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bframe discard;
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# For transmuxing RTMP to RTC, the strategy for aac audio.
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# transcode Transcode aac to opus.
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# discard Discard aac audio packet.
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# default: discard
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aac discard;
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# Whether enable transmuxing RTMP to RTC.
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# If enabled, transcode aac to opus.
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# default: off
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rtmp_to_rtc off;
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# Whether keep B-frame, which is normal feature in live streaming,
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# but usually disabled in RTC.
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# default: off
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keep_bframe off;
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###############################################################
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# For transmuxing RTC to RTMP.
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# Whether trans-mux RTC to RTMP streaming.
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# Whether enable transmuxing RTC to RTMP.
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# Default: off
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rtc_to_rtmp off;
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# The PLI interval in seconds, for RTC to RTMP.
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@ -44,7 +44,7 @@ rtc_server {
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vhost __defaultVhost__ {
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rtc {
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enabled on;
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bframe discard;
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keep_bframe off;
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}
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}
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@ -27,8 +27,8 @@ rtc_server {
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vhost __defaultVhost__ {
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rtc {
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enabled on;
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bframe discard;
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aac transcode;
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rtmp_to_rtc on;
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keep_bframe off;
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rtc_to_rtmp on;
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}
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play {
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@ -30,7 +30,7 @@ vhost __defaultVhost__ {
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rtc {
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enabled on;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
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aac discard;
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rtmp_to_rtc off;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
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rtc_to_rtmp off;
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}
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@ -28,7 +28,7 @@ vhost __defaultVhost__ {
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rtc {
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enabled on;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
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aac transcode;
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rtmp_to_rtc on;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
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rtc_to_rtmp on;
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}
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@ -28,7 +28,7 @@ vhost __defaultVhost__ {
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rtc {
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enabled on;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
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aac transcode;
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rtmp_to_rtc on;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
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rtc_to_rtmp on;
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}
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@ -32,7 +32,7 @@ vhost __defaultVhost__ {
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rtc {
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enabled on;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
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aac discard;
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rtmp_to_rtc off;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
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rtc_to_rtmp off;
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}
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@ -41,7 +41,7 @@ vhost __defaultVhost__ {
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rtc {
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enabled on;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
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aac discard;
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rtmp_to_rtc off;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
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rtc_to_rtmp off;
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}
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