| Index: talk/media/webrtc/webrtcvoiceengine.cc
|
| diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
|
| index 4bd0400dfa8e394642f22a65f77a601b7efec2f7..bd4c47d15bc2d70364e8ac63e934cfe7e9117815 100644
|
| --- a/talk/media/webrtc/webrtcvoiceengine.cc
|
| +++ b/talk/media/webrtc/webrtcvoiceengine.cc
|
| @@ -524,18 +524,6 @@ void WebRtcVoiceEngine::Construct() {
|
| // Load our audio codec list.
|
| codecs_ = WebRtcVoiceCodecs::SupportedCodecs();
|
|
|
| - // Load our RTP Header extensions.
|
| - rtp_header_extensions_.push_back(
|
| - RtpHeaderExtension(kRtpAudioLevelHeaderExtension,
|
| - kRtpAudioLevelHeaderExtensionDefaultId));
|
| - rtp_header_extensions_.push_back(
|
| - RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
|
| - kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
|
| - if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
|
| - rtp_header_extensions_.push_back(RtpHeaderExtension(
|
| - kRtpTransportSequenceNumberHeaderExtension,
|
| - kRtpTransportSequenceNumberHeaderExtensionDefaultId));
|
| - }
|
| options_ = GetDefaultEngineOptions();
|
| }
|
|
|
| @@ -1097,10 +1085,21 @@ const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() {
|
| return codecs_;
|
| }
|
|
|
| -const std::vector<RtpHeaderExtension>&
|
| -WebRtcVoiceEngine::rtp_header_extensions() const {
|
| +std::vector<RtpHeaderExtension>
|
| +WebRtcVoiceEngine::SupportedRtpHeaderExtensions() const {
|
| RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
|
| - return rtp_header_extensions_;
|
| + std::vector<RtpHeaderExtension> rtp_header_extensions;
|
| + rtp_header_extensions.push_back(RtpHeaderExtension(
|
| + kRtpAudioLevelHeaderExtension, kRtpAudioLevelHeaderExtensionDefaultId));
|
| + rtp_header_extensions.push_back(
|
| + RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
|
| + kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
|
| + if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
|
| + rtp_header_extensions.push_back(RtpHeaderExtension(
|
| + kRtpTransportSequenceNumberHeaderExtension,
|
| + kRtpTransportSequenceNumberHeaderExtensionDefaultId));
|
| + }
|
| + return rtp_header_extensions;
|
| }
|
|
|
| int WebRtcVoiceEngine::GetLastEngineError() {
|
|
|