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Unified Diff: webrtc/api/call/audio_send_stream.cc

Issue 2247213005: Fixing config for Audio BWE. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebasing Created 4 years, 2 months ago
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Index: webrtc/api/call/audio_send_stream.cc
diff --git a/webrtc/api/call/audio_send_stream.cc b/webrtc/api/call/audio_send_stream.cc
index 06cbc545d9313846b057ed3432e2862c5c8b9b14..020470fa782d0606bbdf118df267539def2e1ad5 100644
--- a/webrtc/api/call/audio_send_stream.cc
+++ b/webrtc/api/call/audio_send_stream.cc
@@ -39,8 +39,8 @@ std::string AudioSendStream::Config::ToString() const {
ss << "{rtp: " << rtp.ToString();
ss << ", send_transport: " << (send_transport ? "(Transport)" : "nullptr");
ss << ", voe_channel_id: " << voe_channel_id;
- ss << ", min_bitrate_kbps: " << min_bitrate_kbps;
- ss << ", max_bitrate_kbps: " << max_bitrate_kbps;
+ ss << ", min_bitrate_bps: " << min_bitrate_bps;
+ ss << ", max_bitrate_bps: " << max_bitrate_bps;
ss << ", send_codec_spec: " << send_codec_spec.ToString();
ss << '}';
return ss.str();

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