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Unified Diff: webrtc/call/rampup_tests.cc

Issue 2247213005: Fixing config for Audio BWE. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: on Stefan's suggestion Created 4 years, 1 month ago
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Index: webrtc/call/rampup_tests.cc
diff --git a/webrtc/call/rampup_tests.cc b/webrtc/call/rampup_tests.cc
index df64f41051b57dfccf91cf2f06aa6536d77b6196..3cdcc991d2517265121b4474e8f24deb7fdcaea6 100644
--- a/webrtc/call/rampup_tests.cc
+++ b/webrtc/call/rampup_tests.cc
@@ -212,8 +212,8 @@ void RampUpTester::ModifyAudioConfigs(
send_config->rtp.ssrc = audio_ssrcs_[0];
send_config->rtp.extensions.clear();
- send_config->min_bitrate_kbps = 6;
- send_config->max_bitrate_kbps = 60;
+ send_config->min_bitrate_bps = 6000;
+ send_config->max_bitrate_bps = 60000;
bool transport_cc = false;
if (extension_type_ == RtpExtension::kAbsSendTimeUri) {
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