| Index: webrtc/media/engine/webrtcvoiceengine.cc
|
| diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
|
| index 1f197a33e8d4b7261bf52531e686b4e5598e93d9..0fef4fd0a8aa66ee8133193644a86da3ea003cf7 100644
|
| --- a/webrtc/media/engine/webrtcvoiceengine.cc
|
| +++ b/webrtc/media/engine/webrtcvoiceengine.cc
|
| @@ -181,9 +181,10 @@ rtc::Optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
|
| const webrtc::AudioCodecSpec& spec) {
|
| // If application-configured bitrate is set, take minimum of that and SDP
|
| // bitrate.
|
| - const int bps = rtp_max_bitrate_bps
|
| - ? MinPositive(max_send_bitrate_bps, *rtp_max_bitrate_bps)
|
| - : max_send_bitrate_bps;
|
| + const int bps =
|
| + rtp_max_bitrate_bps
|
| + ? webrtc::MinPositive(max_send_bitrate_bps, *rtp_max_bitrate_bps)
|
| + : max_send_bitrate_bps;
|
| if (bps <= 0) {
|
| return rtc::Optional<int>(spec.info.default_bitrate_bps);
|
| }
|
|
|