webrtc/api/call/audio_send_stream.h - Issue 2247213005: Fixing config for Audio BWE.

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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: rebasing Created 4 years, 2 months 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 7ff791e62ad307782382b1243429f971733bdb0f..76ac950612494bc5cba95c9f04302ad264808f55 100644

--- a/webrtc/api/call/audio_send_stream.h

+++ b/webrtc/api/call/audio_send_stream.h

@@ -89,8 +89,8 @@ 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;

struct SendCodecSpec {

SendCodecSpec();