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}
26 lines
814 B
Plaintext
26 lines
814 B
Plaintext
# Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../../webrtc.gni")
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if (rtc_include_tests) {
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rtc_library("pc_scenario_tests") {
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testonly = true
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sources = [ "goog_cc_test.cc" ]
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deps = [
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"../../api:rtc_stats_api",
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"../../modules/rtp_rtcp:rtp_rtcp",
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"../../pc:pc_test_utils",
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"../../pc:rtc_pc_base",
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"../../test:field_trial",
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"../../test:test_support",
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"../../test/peer_scenario:peer_scenario",
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]
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}
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}
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