diff options
| -rw-r--r-- | indra/llwebrtc/llwebrtc.cpp | 87 |
1 files changed, 56 insertions, 31 deletions
diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp index b02354cbac..3152e1eef6 100644 --- a/indra/llwebrtc/llwebrtc.cpp +++ b/indra/llwebrtc/llwebrtc.cpp @@ -106,6 +106,7 @@ void LLWebRTCImpl::init() std::unique_ptr<webrtc::AudioDeviceDataObserver>(mAudioDeviceObserver)); mDeviceModule->Init(); mDeviceModule->SetStereoRecording(false); + mDeviceModule->SetStereoPlayout(true); mDeviceModule->EnableBuiltInAEC(false); updateDevices(); }); @@ -321,6 +322,8 @@ bool LLWebRTCImpl::initializeConnectionThreaded() apm_config.noise_suppression.enabled = true; apm_config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kVeryHigh; apm_config.transient_suppression.enabled = true; + apm_config.pipeline.multi_channel_render = true; + apm_config.pipeline.multi_channel_capture = true; // apm->ApplyConfig(apm_config); @@ -416,8 +419,8 @@ bool LLWebRTCImpl::initializeConnectionThreaded() params.codecs.push_back(codecparam); receiver->SetParameters(params); } - - mPeerConnection->SetLocalDescription(rtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface>(this)); + webrtc::PeerConnectionInterface::RTCOfferAnswerOptions offerOptions; + mPeerConnection->CreateOffer(this, offerOptions); RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->signaling_state(); @@ -516,6 +519,16 @@ void LLWebRTCImpl::OnAddTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>> &streams) { RTC_LOG(LS_INFO) << __FUNCTION__ << " " << receiver->id(); + webrtc::RtpParameters params; + webrtc::RtpCodecParameters codecparam; + codecparam.name = "opus"; + codecparam.kind = cricket::MEDIA_TYPE_AUDIO; + codecparam.clock_rate = 48000; + codecparam.num_channels = 2; + codecparam.parameters["stereo"] = "1"; + codecparam.parameters["sprop-stereo"] = "1"; + params.codecs.push_back(codecparam); + receiver->SetParameters(params); } void LLWebRTCImpl::OnRemoveTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver) @@ -632,11 +645,47 @@ void LLWebRTCImpl::OnSuccess(webrtc::SessionDescriptionInterface *desc) std::string sdp; desc->ToString(&sdp); RTC_LOG(LS_INFO) << sdp; +; + // mangle the sdp as this is the only way currently to bump up + // the send audio rate to 48k + std::istringstream sdp_stream(sdp); + std::ostringstream sdp_mangled_stream; + std::string sdp_line; + int opus_payload = 0; + while (std::getline(sdp_stream, sdp_line)) + { + int bandwidth = 0; + int payload_id = 0; + // force mono down, stereo up + if (std::sscanf(sdp_line.c_str(), "a=rtpmap:%i opus/%i/2", &payload_id, &bandwidth) == 2) + { + sdp_mangled_stream << sdp_line << "\n"; + opus_payload = payload_id; + } + else if (sdp_line.rfind(std::format("a=fmtp:{}", opus_payload)) == 0) + { + sdp_mangled_stream << sdp_line << "a=fmtp:" << opus_payload + << " minptime=10;useinbandfec=1;stereo=1;sprop-stereo=1;maxplaybackrate=48000\n"; + } + else + { + sdp_mangled_stream << sdp_line << "\n"; + } + } + + webrtc::CreateSessionDescription(webrtc::SdpType::kOffer, sdp_mangled_stream.str()); + + + + mPeerConnection->SetLocalDescription(std::unique_ptr<webrtc::SessionDescriptionInterface>( + webrtc::CreateSessionDescription(webrtc::SdpType::kOffer, sdp_mangled_stream.str())), + rtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface>(this)); + RTC_LOG(LS_INFO) << __FUNCTION__ << " Local SDP: " << sdp_mangled_stream.str(); + - RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->signaling_state(); for (auto &observer : mSignalingObserverList) { - observer->OnOfferAvailable(sdp); + observer->OnOfferAvailable(sdp_mangled_stream.str()); } } @@ -669,36 +718,12 @@ void LLWebRTCImpl::OnSetLocalDescriptionComplete(webrtc::RTCError error) auto desc = mPeerConnection->pending_local_description(); std::string sdp; desc->ToString(&sdp); - // mangle the sdp as this is the only way currently to bump up - // the send audio rate to 48k - std::istringstream sdp_stream(sdp); - std::ostringstream sdp_mangled_stream; - std::string sdp_line; - char opus_payload[10]; - while (std::getline(sdp_stream, sdp_line)) { - int bandwidth = 0; - int payload_id = 0; - // force mono down, stereo up - if (std::sscanf(sdp_line.c_str(), "a=rtpmap:%i opus/%i/2", &payload_id, &bandwidth) == 2) - { - sdp_mangled_stream << sdp_line << "\n"; - sprintf(opus_payload,"%d",payload_id); - } - else if (sdp_line.rfind(std::string("a=fmtp:") + opus_payload) == 0) - { - sdp_mangled_stream << sdp_line << "a=fmtp:" << opus_payload - << " stereo=1;sprop-stereo=0;minptime=10;useinbandfec=1;maxplaybackrate=48000\n"; - } - else - { - sdp_mangled_stream << sdp_line << "\n"; - } - } - RTC_LOG(LS_INFO) << __FUNCTION__ << " Local SDP: " << sdp_mangled_stream.str(); + + RTC_LOG(LS_INFO) << __FUNCTION__ << " Local SDP: " << sdp; ; for (auto &observer : mSignalingObserverList) { - observer->OnOfferAvailable(sdp_mangled_stream.str()); + observer->OnOfferAvailable(sdp); } } |