livie
/
phoenix-firestorm
1
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity

add stereo support

meow-7.2.2
Roxie Linden 2023-09-27 18:16:57 -07:00
parent 861f21957e
commit e147381eb3
1 changed files with 56 additions and 31 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

87
indra/llwebrtc/llwebrtc.cpp
View File

@ -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);
}
}