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

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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: on Stefan's suggestion Created 4 years, 1 month 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 52c30f0987b9efe23ee04821dd2dc9db4e783027..3ce35e7a4b865b6019a1f638ddc2d54e2ea1f0c9 100644

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

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

@@ -41,8 +41,8 @@ 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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