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WebRTC: Support config the bitrate of transcoding AAC to Opus. v5.0.167, v6.0.60 (#3515)

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Co-authored-by: john <hondaxiao@tencent.com>
This commit is contained in:
chundonglinlin 2023-07-18 11:09:50 +08:00 • committed by winlin
parent 497ea2bc90
commit e11b93d664
6 changed files with 86 additions and 4 deletions
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10
trunk/conf/full.conf
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@ -546,6 +546,11 @@ vhost rtc.vhost.srs.com {
# Overwrite by env SRS_VHOST_RTC_KEEP_BFRAME for all vhosts.
# default: off
keep_bframe off;
# The transcode audio bitrate, for RTMP to RTC.
# Overwrite by env SRS_VHOST_RTC_OPUS_BITRATE for all vhosts.
# [8000, 320000]
# default: 48000
opus_bitrate 48000;
###############################################################
# Whether enable transmuxing RTC to RTMP.
# Overwrite by env SRS_VHOST_RTC_RTC_TO_RTMP for all vhosts.
@ -556,6 +561,11 @@ vhost rtc.vhost.srs.com {
# Overwrite by env SRS_VHOST_RTC_PLI_FOR_RTMP for all vhosts.
# Default: 6.0
pli_for_rtmp 6.0;
# The transcode audio bitrate, for RTC to RTMP.
# Overwrite by env SRS_VHOST_RTC_AAC_BITRATE for all vhosts.
# [8000, 320000]
# default: 48000
aac_bitrate 48000;
}
###############################################################
# For transmuxing RTMP to RTC, it will impact the default values if RTC is on.