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![]() The decision to route audio packets to a separate overuse detector is off by default and requires the field trial WebRTC-Bwe-SeparateAudioPackets/enabled,packet_threshold:10,time_threshold:1000ms/ The parameters control the threshold for switching over to the audio overuse detector if we stop receiving feedback for video. Bug: webrtc:10932 Change-Id: Icdde35bc7a98b18b1a344bd2d620a890fd9421d9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168342 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30694} |
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.. | ||
media | ||
rtp | ||
test | ||
bitrate_settings.cc | ||
bitrate_settings.h | ||
BUILD.gn | ||
congestion_control_interface.h | ||
data_channel_transport_interface.h | ||
datagram_transport_interface.h | ||
DEPS | ||
enums.h | ||
field_trial_based_config.cc | ||
field_trial_based_config.h | ||
goog_cc_factory.cc | ||
goog_cc_factory.h | ||
network_control.h | ||
network_types.cc | ||
network_types.h | ||
OWNERS | ||
stun.cc | ||
stun.h | ||
stun_unittest.cc | ||
webrtc_key_value_config.h |