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

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/api/call/audio_send_stream.h
diff --git a/webrtc/api/call/audio_send_stream.h b/webrtc/api/call/audio_send_stream.h
index 78ab8ec52e6ea32ab7e8f590a16fa67fb5477c94..658c9de37165a06e200b442e7417f6bc1620a983 100644
--- a/webrtc/api/call/audio_send_stream.h
+++ b/webrtc/api/call/audio_send_stream.h
@@ -90,8 +90,8 @@ class AudioSendStream {
// Bitrate limits used for variable audio bitrate streams. Set both to -1 to
// disable audio bitrate adaptation.
// Note: This is still an experimental feature and not ready for real usage.
- int min_bitrate_kbps = -1;
- int max_bitrate_kbps = -1;
+ int min_bitrate_bps = -1;
+ int max_bitrate_bps = -1;
// Defines whether to turn on audio network adaptor, and defines its config
// string.
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