| 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.
|
|
|