Index: webrtc/media/engine/webrtcvoiceengine.cc |
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc |
index e0d9884e8f1b2939b777b9f61df6e4722636e9ce..d774f788b75cfc15bba3a04d26dde6a20a3a39c1 100644 |
--- a/webrtc/media/engine/webrtcvoiceengine.cc |
+++ b/webrtc/media/engine/webrtcvoiceengine.cc |
@@ -982,8 +982,7 @@ RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const { |
capabilities.header_extensions.push_back( |
webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, |
webrtc::RtpExtension::kAudioLevelDefaultId)); |
- if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") == |
- "Enabled") { |
+ if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) { |
capabilities.header_extensions.push_back(webrtc::RtpExtension( |
webrtc::RtpExtension::kTransportSequenceNumberUri, |
webrtc::RtpExtension::kTransportSequenceNumberDefaultId)); |
@@ -1194,8 +1193,8 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream |
: voe_audio_transport_(voe_audio_transport), |
call_(call), |
config_(send_transport), |
- send_side_bwe_with_overhead_(webrtc::field_trial::FindFullName( |
- "WebRTC-SendSideBwe-WithOverhead") == "Enabled"), |
+ send_side_bwe_with_overhead_( |
+ webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")), |
max_send_bitrate_bps_(max_send_bitrate_bps), |
rtp_parameters_(CreateRtpParametersWithOneEncoding()) { |
RTC_DCHECK_GE(ch, 0); |
@@ -1422,8 +1421,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream |
stream_ = nullptr; |
} |
RTC_DCHECK(!stream_); |
- if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") == |
- "Enabled") { |
+ if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) { |
config_.min_bitrate_bps = kOpusMinBitrateBps; |
config_.max_bitrate_bps = kOpusBitrateFbBps; |
// TODO(mflodman): Keep testing this and set proper values. |