Test field trial group with startswith rather than equals.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 964 matching lines @@
   RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
   return recv_codecs_;
 }
 
 RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
   RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
   RtpCapabilities capabilities;
   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));
   }
   return capabilities;
 }
 
 int WebRtcVoiceEngine::GetLastEngineError() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   return voe_wrapper_->error();
@@ skipping 190 matching lines @@
       const std::string& c_name,
       const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec,
       const std::vector<webrtc::RtpExtension>& extensions,
       int max_send_bitrate_bps,
       const rtc::Optional<std::string>& audio_network_adaptor_config,
       webrtc::Call* call,
       webrtc::Transport* send_transport)
       : 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);
     // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore:
     // RTC_DCHECK(voe_audio_transport);
     RTC_DCHECK(call);
     config_.rtp.ssrc = ssrc;
     config_.rtp.c_name = c_name;
     config_.voe_channel_id = ch;
     config_.rtp.extensions = extensions;
@@ skipping 206 matching lines @@
     }
   }
 
   void RecreateAudioSendStream() {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     if (stream_) {
       call_->DestroyAudioSendStream(stream_);
       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.
       // Note: This is an early experiment currently only supported by Opus.
       if (send_side_bwe_with_overhead_) {
         auto packet_sizes_ms = WebRtcVoiceCodecs::GetPacketSizesMs(
             config_.send_codec_spec.codec_inst);
         if (!packet_sizes_ms.empty()) {
           int max_packet_size_ms =
               *std::max_element(packet_sizes_ms.begin(), packet_sizes_ms.end());
@@ skipping 1225 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE