[WebRTC] Rework device handling sequence so that we can handle unplugging/re-plugging devices (#4593)

* [WebRTC] Rework device handling sequence so that we can handle unplugging/re-plugging devices

The device handling was not processing device updates in the proper sequence as
things like AEC use both input and output devices.  Devices like headsets are both
so unplugging them resulted in various mute conditions and sometimes even a crash.

Now, we update both capture and render devices at once in the proper sequence.

Test Guidance:
* Bring two users in the same place in webrtc regions.
* The 'listening' one should have a headset or something set oas 'Default'
* Press 'talk' on one, and verify the other can hear.
* Unplug the headset from the listening one.
* Validate that audio changes from the headset to the speakers.
* Plug the headset back in.
* Validate that audio changes from speakers to headset.
* Do the same type of test with the headset viewer talking.
* The microphone used should switch from the headset to the computer (it should have one)

Do other various device tests, such as setting devices explicitly, messing with the device selector, etc.

* Fix race condition when multiple change device requests might come in at once

* Update to m137

The primary feature of this commit is to update libwebrtc from m114
to m137.  This is needed to make webrtc buildable, as m114 is not buildable
by the current toolset.

m137 had some changes to the API, which required renaming or changing namespace
of some of the calls.

Additionally, this PR moves from a callback mechanism for gathering the energy
levels for tuning to a wrapper AudioDeviceModule, which gives us more control
over the audio stream.

Finally, the new m137-based webrtc has been updated to allow for 192khz audio
streams.

* Properly pass the observer setting into the inner audio device module

* Update to m137 and get rid of some noise

    This change updates to m137 from m114, which required a few API changes.

    Additionally, this fixes the hiss that happens shortly after someone unmutes: https://github.com/secondlife/server/issues/2094

    There was also an issue with a slight amount of repeated after unmuting if there was audio right before unmuting.  This is because
    the audio processing and buffering still had audio from the previous speaking session.  Now, we inject nearly a half second of
    silence into the audio buffers/processor after unmuting to flush things.

* Install nsis on windows

* Use the newer digital AGC pipeline

m137 improved the AGC pipeline and the existing analog style is going away
so move to the new digital pipeline.

Also, some tweaking for audio levels so that we don't see inworld bars when tuning,
so one's own bars seem a reasonable size, etc.

* Install NSIS during windows sisgning and package build step

* Try pinning the packaging to windows 2022 to deal with missing nsis

* Adjust gain calculation and audio level calculations for tuning and peer connections

* Update with mac universal webrtc build

* Tuning of voice indicators for both tuning mode and inworld for self.

* Redo device deployment to handle cases where multiple deploy requests pile up

Also, mute when leaving webrtc-enabled regions or parcels,
and unmute when voice comes back.

* pre commit issue

1 files changed, 398 insertions, 342 deletions

diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp
index 20951ff816..edba2bee9a 100644
--- a/indra/llwebrtc/llwebrtc.cpp
+++ b/indra/llwebrtc/llwebrtc.cpp
@@ -9,7 +9,7 @@
  * This library is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation;
- * version 2.1 of the License only.
+ * version 2.1 of the License only
  *
  * This library is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
@@ -32,41 +32,79 @@
 #include "api/audio_codecs/audio_encoder_factory.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/audio/builtin_audio_processing_builder.h"
 #include "api/media_stream_interface.h"
 #include "api/media_stream_track.h"
 #include "modules/audio_processing/audio_buffer.h"
 #include "modules/audio_mixer/audio_mixer_impl.h"
+#include "api/environment/environment_factory.h"
 
 namespace llwebrtc
 {
+#if WEBRTC_WIN
+static int16_t PLAYOUT_DEVICE_DEFAULT = webrtc::AudioDeviceModule::kDefaultDevice;
+static int16_t RECORD_DEVICE_DEFAULT  = webrtc::AudioDeviceModule::kDefaultDevice;
+#else
+static int16_t PLAYOUT_DEVICE_DEFAULT = 0;
+static int16_t RECORD_DEVICE_DEFAULT  = 0;
+#endif
 
-static int16_t PLAYOUT_DEVICE_DEFAULT = -1;
-static int16_t PLAYOUT_DEVICE_BAD     = -2;
-static int16_t RECORD_DEVICE_DEFAULT  = -1;
-static int16_t RECORD_DEVICE_BAD      = -2;
 
-LLAudioDeviceObserver::LLAudioDeviceObserver() : mSumVector {0}, mMicrophoneEnergy(0.0) {}
+//
+// LLWebRTCAudioTransport implementation
+//
 
-float LLAudioDeviceObserver::getMicrophoneEnergy() { return mMicrophoneEnergy; }
+LLWebRTCAudioTransport::LLWebRTCAudioTransport() : mMicrophoneEnergy(0.0)
+{
+    memset(mSumVector, 0, sizeof(mSumVector));
+}
 
-// TODO: Pull smoothing/filtering code into a common helper function
-// for LLAudioDeviceObserver and LLCustomProcessor
+void LLWebRTCAudioTransport::SetEngineTransport(webrtc::AudioTransport* t)
+{
+    engine_.store(t, std::memory_order_release);
+}
 
-void LLAudioDeviceObserver::OnCaptureData(const void    *audio_samples,
-                                          const size_t   num_samples,
-                                          const size_t   bytes_per_sample,
-                                          const size_t   num_channels,
-                                          const uint32_t samples_per_sec)
+int32_t LLWebRTCAudioTransport::RecordedDataIsAvailable(const void* audio_data,
+                                                        size_t      number_of_frames,
+                                                        size_t      bytes_per_frame,
+                                                        size_t      number_of_channels,
+                                                        uint32_t    samples_per_sec,
+                                                        uint32_t    total_delay_ms,
+                                                        int32_t     clock_drift,
+                                                        uint32_t    current_mic_level,
+                                                        bool        key_pressed,
+                                                        uint32_t&   new_mic_level)
 {
+    auto* engine = engine_.load(std::memory_order_acquire);
+
+    // 1) Deliver to engine (authoritative).
+    int32_t ret = 0;
+    if (engine)
+    {
+        ret = engine->RecordedDataIsAvailable(audio_data,
+                                              number_of_frames,
+                                              bytes_per_frame,
+                                              number_of_channels,
+                                              samples_per_sec,
+                                              total_delay_ms,
+                                              clock_drift,
+                                              current_mic_level,
+                                              key_pressed,
+                                              new_mic_level);
+    }
+
+    // 2) Calculate energy for microphone level monitoring
     // calculate the energy
     float        energy  = 0;
-    const short *samples = (const short *) audio_samples;
-    for (size_t index = 0; index < num_samples * num_channels; index++)
+    const short *samples = (const short *) audio_data;
+
+    for (size_t index = 0; index < number_of_frames * number_of_channels; index++)
     {
         float sample = (static_cast<float>(samples[index]) / (float) 32767);
         energy += sample * sample;
     }
-
+    float gain = mGain.load(std::memory_order_relaxed);
+    energy     = energy * gain * gain;
     // smooth it.
     size_t buffer_size = sizeof(mSumVector) / sizeof(mSumVector[0]);
     float  totalSum    = 0;
@@ -78,18 +116,59 @@ void LLAudioDeviceObserver::OnCaptureData(const void    *audio_samples,
     }
     mSumVector[i] = energy;
     totalSum += energy;
-    mMicrophoneEnergy = std::sqrt(totalSum / (num_samples * buffer_size));
+    mMicrophoneEnergy = std::sqrt(totalSum / (number_of_frames * number_of_channels * buffer_size));
+
+    return ret;
 }
 
-void LLAudioDeviceObserver::OnRenderData(const void    *audio_samples,
-                                         const size_t   num_samples,
-                                         const size_t   bytes_per_sample,
-                                         const size_t   num_channels,
-                                         const uint32_t samples_per_sec)
+int32_t LLWebRTCAudioTransport::NeedMorePlayData(size_t   number_of_frames,
+                                                 size_t   bytes_per_frame,
+                                                 size_t   number_of_channels,
+                                                 uint32_t samples_per_sec,
+                                                 void*    audio_data,
+                                                 size_t&  number_of_samples_out,
+                                                 int64_t* elapsed_time_ms,
+                                                 int64_t* ntp_time_ms)
 {
+    auto* engine = engine_.load(std::memory_order_acquire);
+    if (!engine)
+    {
+        // No engine sink; output silence to be safe.
+        const size_t bytes = number_of_frames * bytes_per_frame * number_of_channels;
+        memset(audio_data, 0, bytes);
+        number_of_samples_out = bytes_per_frame;
+        return 0;
+    }
+
+    // Only the engine should fill the buffer.
+    return engine->NeedMorePlayData(number_of_frames,
+                                    bytes_per_frame,
+                                    number_of_channels,
+                                    samples_per_sec,
+                                    audio_data,
+                                    number_of_samples_out,
+                                    elapsed_time_ms,
+                                    ntp_time_ms);
 }
 
-LLCustomProcessor::LLCustomProcessor() : mSampleRateHz(0), mNumChannels(0), mMicrophoneEnergy(0.0), mGain(1.0)
+void LLWebRTCAudioTransport::PullRenderData(int      bits_per_sample,
+                                            int      sample_rate,
+                                            size_t   number_of_channels,
+                                            size_t   number_of_frames,
+                                            void*    audio_data,
+                                            int64_t* elapsed_time_ms,
+                                            int64_t* ntp_time_ms)
+{
+    auto* engine = engine_.load(std::memory_order_acquire);
+
+    if (engine)
+    {
+        engine
+            ->PullRenderData(bits_per_sample, sample_rate, number_of_channels, number_of_frames, audio_data, elapsed_time_ms, ntp_time_ms);
+    }
+}
+
+LLCustomProcessor::LLCustomProcessor(LLCustomProcessorStatePtr state) : mSampleRateHz(0), mNumChannels(0), mState(state)
 {
     memset(mSumVector, 0, sizeof(mSumVector));
 }
@@ -101,40 +180,61 @@ void LLCustomProcessor::Initialize(int sample_rate_hz, int num_channels)
     memset(mSumVector, 0, sizeof(mSumVector));
 }
 
-void LLCustomProcessor::Process(webrtc::AudioBuffer *audio_in)
+void LLCustomProcessor::Process(webrtc::AudioBuffer *audio)
 {
-    webrtc::StreamConfig stream_config;
-    stream_config.set_sample_rate_hz(mSampleRateHz);
-    stream_config.set_num_channels(mNumChannels);
-    std::vector<float *> frame;
-    std::vector<float>   frame_samples;
-
-    if (audio_in->num_channels() < 1 || audio_in->num_frames() < 480)
+    if (audio->num_channels() < 1 || audio->num_frames() < 480)
     {
         return;
     }
 
-    // grab the input audio
-    frame_samples.resize(stream_config.num_samples());
-    frame.resize(stream_config.num_channels());
-    for (size_t ch = 0; ch < stream_config.num_channels(); ++ch)
+    // calculate the energy
+
+    float desired_gain = mState->getGain();
+    if (mState->getDirty())
     {
-        frame[ch] = &(frame_samples)[ch * stream_config.num_frames()];
+        // We'll delay ramping by 30ms in order to clear out buffers that may
+        // have had content before muting.  And for the last 20ms, we'll ramp
+        // down or up smoothly.
+        mRampFrames = 5;
+
+        // we've changed our desired gain, so set the incremental
+        // gain change so that we smoothly step over 20ms
+        mGainStep = (desired_gain - mCurrentGain) / (mSampleRateHz / 50);
     }
 
-    audio_in->CopyTo(stream_config, &frame[0]);
-
-    // calculate the energy
-    float energy = 0;
-    for (size_t index = 0; index < stream_config.num_samples(); index++)
+    if (mRampFrames)
     {
-        float sample = frame_samples[index];
-        sample       = sample * mGain; // apply gain
-        frame_samples[index] = sample; // write processed sample back to buffer.
-        energy += sample * sample;
+        if (mRampFrames-- > 2)
+        {
+            // don't change the gain if we're still in the 'don't move' phase
+            mGainStep = 0.0f;
+        }
+    }
+    else
+    {
+        // We've ramped all the way down, so don't step the gain any more and
+        // just maintaint he current gain.
+        mGainStep = 0.0f;
+        mCurrentGain = desired_gain;
     }
 
-    audio_in->CopyFrom(&frame[0], stream_config);
+    float energy       = 0;
+
+    auto chans = audio->channels();
+    for (size_t ch = 0; ch < audio->num_channels(); ch++)
+    {
+        float* frame_samples = chans[ch];
+        float  gain          = mCurrentGain;
+        for (size_t index = 0; index < audio->num_frames(); index++)
+        {
+            float sample         = frame_samples[index];
+            sample               = sample * gain;    // apply gain
+            frame_samples[index] = sample;        // write processed sample back to buffer.
+            energy += sample * sample;
+            gain += mGainStep;
+        }
+    }
+    mCurrentGain += audio->num_frames() * mGainStep;
 
     // smooth it.
     size_t buffer_size = sizeof(mSumVector) / sizeof(mSumVector[0]);
@@ -147,7 +247,7 @@ void LLCustomProcessor::Process(webrtc::AudioBuffer *audio_in)
     }
     mSumVector[i] = energy;
     totalSum += energy;
-    mMicrophoneEnergy = std::sqrt(totalSum / (stream_config.num_samples() * buffer_size));
+    mState->setMicrophoneEnergy(std::sqrt(totalSum / (audio->num_channels() * audio->num_frames() * buffer_size)));
 }
 
 //
@@ -159,89 +259,54 @@ LLWebRTCImpl::LLWebRTCImpl(LLWebRTCLogCallback* logCallback) :
     mPeerCustomProcessor(nullptr),
     mMute(true),
     mTuningMode(false),
-    mPlayoutDevice(0),
-    mRecordingDevice(0),
-    mTuningAudioDeviceObserver(nullptr)
+    mDevicesDeploying(0),
+    mGain(0.0f)
 {
 }
 
 void LLWebRTCImpl::init()
 {
-    mPlayoutDevice   = 0;
-    mRecordingDevice = 0;
-    rtc::InitializeSSL();
+    webrtc::InitializeSSL();
 
     // Normal logging is rather spammy, so turn it off.
-    rtc::LogMessage::LogToDebug(rtc::LS_NONE);
-    rtc::LogMessage::SetLogToStderr(true);
-    rtc::LogMessage::AddLogToStream(mLogSink, rtc::LS_VERBOSE);
+    webrtc::LogMessage::LogToDebug(webrtc::LS_NONE);
+    webrtc::LogMessage::SetLogToStderr(true);
+    webrtc::LogMessage::AddLogToStream(mLogSink, webrtc::LS_VERBOSE);
 
     mTaskQueueFactory = webrtc::CreateDefaultTaskQueueFactory();
 
     // Create the native threads.
-    mNetworkThread = rtc::Thread::CreateWithSocketServer();
+    mNetworkThread = webrtc::Thread::CreateWithSocketServer();
     mNetworkThread->SetName("WebRTCNetworkThread", nullptr);
     mNetworkThread->Start();
-    mWorkerThread = rtc::Thread::Create();
+    mWorkerThread = webrtc::Thread::Create();
     mWorkerThread->SetName("WebRTCWorkerThread", nullptr);
     mWorkerThread->Start();
-    mSignalingThread = rtc::Thread::Create();
+    mSignalingThread = webrtc::Thread::Create();
     mSignalingThread->SetName("WebRTCSignalingThread", nullptr);
     mSignalingThread->Start();
 
-    mTuningAudioDeviceObserver = new LLAudioDeviceObserver;
-    mWorkerThread->PostTask(
-        [this]()
-        {
-            // Initialize the audio devices on the Worker Thread
-            mTuningDeviceModule =
-                webrtc::CreateAudioDeviceWithDataObserver(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
-                                                          mTaskQueueFactory.get(),
-                                                          std::unique_ptr<webrtc::AudioDeviceDataObserver>(mTuningAudioDeviceObserver));
-
-            mTuningDeviceModule->Init();
-            mTuningDeviceModule->SetPlayoutDevice(mPlayoutDevice);
-            mTuningDeviceModule->SetRecordingDevice(mRecordingDevice);
-            mTuningDeviceModule->EnableBuiltInAEC(false);
-            mTuningDeviceModule->SetAudioDeviceSink(this);
-            mTuningDeviceModule->InitMicrophone();
-            mTuningDeviceModule->InitSpeaker();
-            mTuningDeviceModule->SetStereoRecording(false);
-            mTuningDeviceModule->SetStereoPlayout(true);
-            mTuningDeviceModule->InitRecording();
-            mTuningDeviceModule->InitPlayout();
-            updateDevices();
-        });
-
     mWorkerThread->BlockingCall(
         [this]()
         {
-            // the peer device module doesn't need an observer
-            // as we pull peer data after audio processing.
-            mPeerDeviceModule = webrtc::CreateAudioDeviceWithDataObserver(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
-                                                                          mTaskQueueFactory.get(),
-                                                                          nullptr);
-            mPeerDeviceModule->Init();
-            mPeerDeviceModule->SetPlayoutDevice(mPlayoutDevice);
-            mPeerDeviceModule->SetRecordingDevice(mRecordingDevice);
-            mPeerDeviceModule->EnableBuiltInAEC(false);
-            mPeerDeviceModule->InitMicrophone();
-            mPeerDeviceModule->InitSpeaker();
+            webrtc::scoped_refptr<webrtc::AudioDeviceModule> realADM =
+                webrtc::AudioDeviceModule::Create(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio, mTaskQueueFactory.get());
+            mDeviceModule = webrtc::make_ref_counted<LLWebRTCAudioDeviceModule>(realADM);
+            mDeviceModule->SetObserver(this);
         });
 
     // The custom processor allows us to retrieve audio data (and levels)
     // from after other audio processing such as AEC, AGC, etc.
-    mPeerCustomProcessor = new LLCustomProcessor;
-    webrtc::AudioProcessingBuilder apb;
-    apb.SetCapturePostProcessing(std::unique_ptr<webrtc::CustomProcessing>(mPeerCustomProcessor));
-    mAudioProcessingModule = apb.Create();
+    mPeerCustomProcessor = std::make_shared<LLCustomProcessorState>();
+    webrtc::BuiltinAudioProcessingBuilder apb;
+    apb.SetCapturePostProcessing(std::make_unique<LLCustomProcessor>(mPeerCustomProcessor));
+    mAudioProcessingModule = apb.Build(webrtc::CreateEnvironment());
 
     webrtc::AudioProcessing::Config apm_config;
     apm_config.echo_canceller.enabled         = false;
     apm_config.echo_canceller.mobile_mode     = false;
     apm_config.gain_controller1.enabled       = false;
-    apm_config.gain_controller1.mode          = webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog;
-    apm_config.gain_controller2.enabled       = false;
+    apm_config.gain_controller2.enabled       = true;
     apm_config.high_pass_filter.enabled       = true;
     apm_config.noise_suppression.enabled      = true;
     apm_config.noise_suppression.level        = webrtc::AudioProcessing::Config::NoiseSuppression::kVeryHigh;
@@ -252,6 +317,7 @@ void LLWebRTCImpl::init()
     mAudioProcessingModule->ApplyConfig(apm_config);
 
     webrtc::ProcessingConfig processing_config;
+
     processing_config.input_stream().set_num_channels(2);
     processing_config.input_stream().set_sample_rate_hz(48000);
     processing_config.output_stream().set_num_channels(2);
@@ -266,13 +332,19 @@ void LLWebRTCImpl::init()
     mPeerConnectionFactory = webrtc::CreatePeerConnectionFactory(mNetworkThread.get(),
                                                                  mWorkerThread.get(),
                                                                  mSignalingThread.get(),
-                                                                 mPeerDeviceModule,
+                                                                 mDeviceModule,
                                                                  webrtc::CreateBuiltinAudioEncoderFactory(),
                                                                  webrtc::CreateBuiltinAudioDecoderFactory(),
                                                                  nullptr /* video_encoder_factory */,
                                                                  nullptr /* video_decoder_factory */,
                                                                  nullptr /* audio_mixer */,
                                                                  mAudioProcessingModule);
+    mWorkerThread->PostTask(
+        [this]()
+        {
+            mDeviceModule->EnableBuiltInAEC(false);
+            updateDevices();
+        });
 
 }
 
@@ -294,64 +366,16 @@ void LLWebRTCImpl::terminate()
     mWorkerThread->BlockingCall(
         [this]()
         {
-            if (mTuningDeviceModule)
-            {
-                mTuningDeviceModule->StopRecording();
-                mTuningDeviceModule->Terminate();
-            }
-            if (mPeerDeviceModule)
+            if (mDeviceModule)
             {
-                mPeerDeviceModule->StopRecording();
-                mPeerDeviceModule->Terminate();
-            }
-            mTuningDeviceModule = nullptr;
-            mPeerDeviceModule   = nullptr;
-            mTaskQueueFactory   = nullptr;
-        });
-    rtc::LogMessage::RemoveLogToStream(mLogSink);
-}
-
-//
-// Devices functions
-//
-// Most device-related functionality needs to happen
-// on the worker thread (the audio thread,) so those calls will be
-// proxied over to that thread.
-//
-void LLWebRTCImpl::setRecording(bool recording)
-{
-    mWorkerThread->PostTask(
-        [this, recording]()
-        {
-            if (recording)
-            {
-                mPeerDeviceModule->SetStereoRecording(false);
-                mPeerDeviceModule->InitRecording();
-                mPeerDeviceModule->StartRecording();
-            }
-            else
-            {
-                mPeerDeviceModule->StopRecording();
-            }
-        });
-}
-
-void LLWebRTCImpl::setPlayout(bool playing)
-{
-    mWorkerThread->PostTask(
-        [this, playing]()
-        {
-            if (playing)
-            {
-                mPeerDeviceModule->SetStereoPlayout(true);
-                mPeerDeviceModule->InitPlayout();
-                mPeerDeviceModule->StartPlayout();
-            }
-            else
-            {
-                mPeerDeviceModule->StopPlayout();
+                mDeviceModule->StopRecording();
+                mDeviceModule->StopPlayout();
+                mDeviceModule->Terminate();
             }
+            mDeviceModule     = nullptr;
+            mTaskQueueFactory = nullptr;
         });
+    webrtc::LogMessage::RemoveLogToStream(mLogSink);
 }
 
 void LLWebRTCImpl::setAudioConfig(LLWebRTCDeviceInterface::AudioConfig config)
@@ -359,9 +383,9 @@ void LLWebRTCImpl::setAudioConfig(LLWebRTCDeviceInterface::AudioConfig config)
     webrtc::AudioProcessing::Config apm_config;
     apm_config.echo_canceller.enabled         = config.mEchoCancellation;
     apm_config.echo_canceller.mobile_mode     = false;
-    apm_config.gain_controller1.enabled       = config.mAGC;
-    apm_config.gain_controller1.mode          = webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog;
-    apm_config.gain_controller2.enabled       = false;
+    apm_config.gain_controller1.enabled       = false;
+    apm_config.gain_controller2.enabled       = config.mAGC;
+    apm_config.gain_controller2.adaptive_digital.enabled = true; // auto-level speech
     apm_config.high_pass_filter.enabled       = true;
     apm_config.transient_suppression.enabled  = true;
     apm_config.pipeline.multi_channel_render  = true;
@@ -414,142 +438,134 @@ void LLWebRTCImpl::unsetDevicesObserver(LLWebRTCDevicesObserver *observer)
     }
 }
 
-void ll_set_device_module_capture_device(rtc::scoped_refptr<webrtc::AudioDeviceModule> device_module, int16_t device)
+// must be run in the worker thread.
+void LLWebRTCImpl::workerDeployDevices()
 {
+    int16_t recordingDevice = RECORD_DEVICE_DEFAULT;
 #if WEBRTC_WIN
-    if (device < 0)
-    {
-        device_module->SetRecordingDevice(webrtc::AudioDeviceModule::kDefaultDevice);
-    }
-    else
-    {
-        device_module->SetRecordingDevice(device);
-    }
+    int16_t recording_device_start = 0;
 #else
-    // passed in default is -1, but the device list
-    // has it at 0
-    device_module->SetRecordingDevice(device + 1);
+    int16_t recording_device_start = 1;
 #endif
-    device_module->InitMicrophone();
-}
 
-void LLWebRTCImpl::setCaptureDevice(const std::string &id)
-{
-    int16_t recordingDevice = RECORD_DEVICE_DEFAULT;
-    if (id != "Default")
+    if (mRecordingDevice != "Default")
     {
-        for (int16_t i = 0; i < mRecordingDeviceList.size(); i++)
+        for (int16_t i = recording_device_start; i < mRecordingDeviceList.size(); i++)
         {
-            if (mRecordingDeviceList[i].mID == id)
+            if (mRecordingDeviceList[i].mID == mRecordingDevice)
             {
                 recordingDevice = i;
                 break;
             }
         }
     }
-    if (recordingDevice == mRecordingDevice)
-    {
-        return;
-    }
-    mRecordingDevice = recordingDevice;
-    if (mTuningMode)
-    {
-        mWorkerThread->PostTask([this, recordingDevice]()
-            {
-                ll_set_device_module_capture_device(mTuningDeviceModule, recordingDevice);
-            });
-    }
-    else
-    {
-        mWorkerThread->PostTask([this, recordingDevice]()
-            {
-                bool recording = mPeerDeviceModule->Recording();
-                if (recording)
-                {
-                    mPeerDeviceModule->StopRecording();
-                }
-                ll_set_device_module_capture_device(mPeerDeviceModule, recordingDevice);
-                if (recording)
-                {
-                    mPeerDeviceModule->SetStereoRecording(false);
-                    mPeerDeviceModule->InitRecording();
-                    mPeerDeviceModule->StartRecording();
-                }
-            });
-    }
-}
 
-
-void ll_set_device_module_render_device(rtc::scoped_refptr<webrtc::AudioDeviceModule> device_module, int16_t device)
-{
+    mDeviceModule->StopPlayout();
+    mDeviceModule->ForceStopRecording();
 #if WEBRTC_WIN
-    if (device < 0)
+    if (recordingDevice < 0)
     {
-        device_module->SetPlayoutDevice(webrtc::AudioDeviceModule::kDefaultDevice);
+        mDeviceModule->SetRecordingDevice((webrtc::AudioDeviceModule::WindowsDeviceType)recordingDevice);
     }
     else
     {
-        device_module->SetPlayoutDevice(device);
+        mDeviceModule->SetRecordingDevice(recordingDevice);
     }
 #else
-    device_module->SetPlayoutDevice(device + 1);
+    mDeviceModule->SetRecordingDevice(recordingDevice);
 #endif
-    device_module->InitSpeaker();
-}
+    mDeviceModule->InitMicrophone();
+    mDeviceModule->SetStereoRecording(false);
+    mDeviceModule->InitRecording();
 
-void LLWebRTCImpl::setRenderDevice(const std::string &id)
-{
     int16_t playoutDevice = PLAYOUT_DEVICE_DEFAULT;
-    if (id != "Default")
+#if WEBRTC_WIN
+    int16_t playout_device_start = 0;
+#else
+    int16_t playout_device_start = 1;
+#endif
+    if (mPlayoutDevice != "Default")
     {
-        for (int16_t i = 0; i < mPlayoutDeviceList.size(); i++)
+        for (int16_t i = playout_device_start; i < mPlayoutDeviceList.size(); i++)
         {
-            if (mPlayoutDeviceList[i].mID == id)
+            if (mPlayoutDeviceList[i].mID == mPlayoutDevice)
             {
                 playoutDevice = i;
                 break;
             }
         }
     }
-    if (playoutDevice == mPlayoutDevice)
+
+#if WEBRTC_WIN
+    if (playoutDevice < 0)
     {
-        return;
+        mDeviceModule->SetPlayoutDevice((webrtc::AudioDeviceModule::WindowsDeviceType)playoutDevice);
+    }
+    else
+    {
+        mDeviceModule->SetPlayoutDevice(playoutDevice);
     }
-    mPlayoutDevice = playoutDevice;
+#else
+    mDeviceModule->SetPlayoutDevice(playoutDevice);
+#endif
+    mDeviceModule->InitSpeaker();
+    mDeviceModule->SetStereoPlayout(true);
+    mDeviceModule->InitPlayout();
 
-    if (mTuningMode)
+    if ((!mMute && mPeerConnections.size()) || mTuningMode)
     {
-        mWorkerThread->PostTask(
-            [this, playoutDevice]()
-            {
-                ll_set_device_module_render_device(mTuningDeviceModule, playoutDevice);
-            });
+        mDeviceModule->ForceStartRecording();
     }
-    else
+
+    if (!mTuningMode)
     {
-        mWorkerThread->PostTask(
-            [this, playoutDevice]()
+        mDeviceModule->StartPlayout();
+    }
+    mSignalingThread->PostTask(
+        [this]
+        {
+            for (auto& connection : mPeerConnections)
             {
-                bool playing = mPeerDeviceModule->Playing();
-                if (playing)
+                if (mTuningMode)
                 {
-                    mPeerDeviceModule->StopPlayout();
+                    connection->enableSenderTracks(false);
                 }
-                ll_set_device_module_render_device(mPeerDeviceModule, playoutDevice);
-                if (playing)
+                else
                 {
-                    mPeerDeviceModule->SetStereoPlayout(true);
-                    mPeerDeviceModule->InitPlayout();
-                    mPeerDeviceModule->StartPlayout();
+                    connection->resetMute();
                 }
-            });
+                connection->enableReceiverTracks(!mTuningMode);
+            }
+            if (1 < mDevicesDeploying.fetch_sub(1, std::memory_order_relaxed))
+            {
+                mWorkerThread->PostTask([this] { workerDeployDevices(); });
+            }
+        });
+}
+
+void LLWebRTCImpl::setCaptureDevice(const std::string &id)
+{
+
+    if (mRecordingDevice != id)
+    {
+        mRecordingDevice = id;
+        deployDevices();
+    }
+}
+
+void LLWebRTCImpl::setRenderDevice(const std::string &id)
+{
+    if (mPlayoutDevice != id)
+    {
+        mPlayoutDevice = id;
+        deployDevices();
     }
 }
 
 // updateDevices needs to happen on the worker thread.
 void LLWebRTCImpl::updateDevices()
 {
-    int16_t renderDeviceCount  = mTuningDeviceModule->PlayoutDevices();
+    int16_t renderDeviceCount  = mDeviceModule->PlayoutDevices();
 
     mPlayoutDeviceList.clear();
 #if WEBRTC_WIN
@@ -563,11 +579,11 @@ void LLWebRTCImpl::updateDevices()
     {
         char name[webrtc::kAdmMaxDeviceNameSize];
         char guid[webrtc::kAdmMaxGuidSize];
-        mTuningDeviceModule->PlayoutDeviceName(index, name, guid);
+        mDeviceModule->PlayoutDeviceName(index, name, guid);
         mPlayoutDeviceList.emplace_back(name, guid);
     }
 
-    int16_t captureDeviceCount        = mTuningDeviceModule->RecordingDevices();
+    int16_t captureDeviceCount        = mDeviceModule->RecordingDevices();
 
     mRecordingDeviceList.clear();
 #if WEBRTC_WIN
@@ -581,7 +597,7 @@ void LLWebRTCImpl::updateDevices()
     {
         char name[webrtc::kAdmMaxDeviceNameSize];
         char guid[webrtc::kAdmMaxGuidSize];
-        mTuningDeviceModule->RecordingDeviceName(index, name, guid);
+        mDeviceModule->RecordingDeviceName(index, name, guid);
         mRecordingDeviceList.emplace_back(name, guid);
     }
 
@@ -593,11 +609,7 @@ void LLWebRTCImpl::updateDevices()
 
 void LLWebRTCImpl::OnDevicesUpdated()
 {
-    // reset these to a bad value so an update is forced
-    mRecordingDevice = RECORD_DEVICE_BAD;
-    mPlayoutDevice   = PLAYOUT_DEVICE_BAD;
-
-    updateDevices();
+    deployDevices();
 }
 
 
@@ -605,60 +617,109 @@ void LLWebRTCImpl::setTuningMode(bool enable)
 {
     mTuningMode = enable;
     mWorkerThread->PostTask(
-        [this, enable] {
-            if (enable)
-            {
-                mPeerDeviceModule->StopRecording();
-                mPeerDeviceModule->StopPlayout();
-                ll_set_device_module_render_device(mTuningDeviceModule, mPlayoutDevice);
-                ll_set_device_module_capture_device(mTuningDeviceModule, mRecordingDevice);
-                mTuningDeviceModule->InitPlayout();
-                mTuningDeviceModule->InitRecording();
-                mTuningDeviceModule->StartRecording();
-                // TODO:  Starting Playout on the TDM appears to create an audio artifact (click)
-                // in this case, so disabling it for now.  We may have to do something different
-                // if we enable 'echo playback' via the TDM when tuning.
-                //mTuningDeviceModule->StartPlayout();
-            }
-            else
-            {
-                mTuningDeviceModule->StopRecording();
-                //mTuningDeviceModule->StopPlayout();
-                ll_set_device_module_render_device(mPeerDeviceModule, mPlayoutDevice);
-                ll_set_device_module_capture_device(mPeerDeviceModule, mRecordingDevice);
-                mPeerDeviceModule->SetStereoPlayout(true);
-                mPeerDeviceModule->SetStereoRecording(false);
-                mPeerDeviceModule->InitPlayout();
-                mPeerDeviceModule->InitRecording();
-                mPeerDeviceModule->StartPlayout();
-                mPeerDeviceModule->StartRecording();
-            }
-        }
-    );
-    mSignalingThread->PostTask(
-        [this, enable]
+        [this]
         {
-            for (auto &connection : mPeerConnections)
-            {
-                if (enable)
+            mDeviceModule->SetTuning(mTuningMode, mMute);
+            mSignalingThread->PostTask(
+                [this]
                 {
-                    connection->enableSenderTracks(false);
-                }
-                else
-                {
-                    connection->resetMute();
-                }
-                connection->enableReceiverTracks(!enable);
-            }
+                    for (auto& connection : mPeerConnections)
+                    {
+                        if (mTuningMode)
+                        {
+                            connection->enableSenderTracks(false);
+                        }
+                        else
+                        {
+                            connection->resetMute();
+                        }
+                        connection->enableReceiverTracks(!mTuningMode);
+                    }
+                });
+        });
+}
+
+void LLWebRTCImpl::deployDevices()
+{
+    if (0 < mDevicesDeploying.fetch_add(1, std::memory_order_relaxed))
+    {
+        return;
+    }
+    mWorkerThread->PostTask(
+        [this] {
+            workerDeployDevices();
         });
 }
 
-float LLWebRTCImpl::getTuningAudioLevel() { return -20 * log10f(mTuningAudioDeviceObserver->getMicrophoneEnergy()); }
+float LLWebRTCImpl::getTuningAudioLevel()
+{
+    return mDeviceModule ? -20 * log10f(mDeviceModule->GetMicrophoneEnergy()) : std::numeric_limits<float>::infinity();
+}
 
-float LLWebRTCImpl::getPeerConnectionAudioLevel() { return -20 * log10f(mPeerCustomProcessor->getMicrophoneEnergy()); }
+void LLWebRTCImpl::setTuningMicGain(float gain)
+{
+    if (mTuningMode && mDeviceModule)
+    {
+        mDeviceModule->SetTuningMicGain(gain);
+    }
+}
+
+float LLWebRTCImpl::getPeerConnectionAudioLevel()
+{
+    return mTuningMode ? std::numeric_limits<float>::infinity()
+                       : (mPeerCustomProcessor ? -20 * log10f(mPeerCustomProcessor->getMicrophoneEnergy())
+                                               : std::numeric_limits<float>::infinity());
+}
 
-void LLWebRTCImpl::setPeerConnectionGain(float gain) { mPeerCustomProcessor->setGain(gain); }
+void LLWebRTCImpl::setMicGain(float gain)
+{
+    mGain = gain;
+    if (!mTuningMode && mPeerCustomProcessor)
+    {
+        mPeerCustomProcessor->setGain(gain);
+    }
+}
 
+void LLWebRTCImpl::setMute(bool mute, int delay_ms)
+{
+    if (mMute != mute)
+    {
+        mMute = mute;
+        intSetMute(mute, delay_ms);
+    }
+}
+
+void LLWebRTCImpl::intSetMute(bool mute, int delay_ms)
+{
+    if (mPeerCustomProcessor)
+    {
+        mPeerCustomProcessor->setGain(mMute ? 0.0f : mGain);
+    }
+    if (mMute)
+    {
+        mWorkerThread->PostDelayedTask(
+            [this]
+            {
+                if (mDeviceModule)
+                {
+                    mDeviceModule->ForceStopRecording();
+                }
+            },
+            webrtc::TimeDelta::Millis(delay_ms));
+    }
+    else
+    {
+        mWorkerThread->PostTask(
+            [this]
+            {
+                if (mDeviceModule)
+                {
+                    mDeviceModule->InitRecording();
+                    mDeviceModule->ForceStartRecording();
+                }
+            });
+    }
+}
 
 //
 // Peer Connection Helpers
@@ -666,34 +727,31 @@ void LLWebRTCImpl::setPeerConnectionGain(float gain) { mPeerCustomProcessor->set
 
 LLWebRTCPeerConnectionInterface *LLWebRTCImpl::newPeerConnection()
 {
-    rtc::scoped_refptr<LLWebRTCPeerConnectionImpl> peerConnection = rtc::scoped_refptr<LLWebRTCPeerConnectionImpl>(new rtc::RefCountedObject<LLWebRTCPeerConnectionImpl>());
+    bool empty = mPeerConnections.empty();
+    webrtc::scoped_refptr<LLWebRTCPeerConnectionImpl> peerConnection = webrtc::scoped_refptr<LLWebRTCPeerConnectionImpl>(new webrtc::RefCountedObject<LLWebRTCPeerConnectionImpl>());
     peerConnection->init(this);
-
-    mPeerConnections.emplace_back(peerConnection);
-    // Should it really start disabled?
-    // Seems like something doesn't get the memo and senders need to be reset later
-    // to remove the voice indicator from taskbar
-    peerConnection->enableSenderTracks(false);
     if (mPeerConnections.empty())
     {
-        setRecording(true);
-        setPlayout(true);
+        intSetMute(mMute);
     }
+    mPeerConnections.emplace_back(peerConnection);
+
+    peerConnection->enableSenderTracks(false);
+    peerConnection->resetMute();
     return peerConnection.get();
 }
 
 void LLWebRTCImpl::freePeerConnection(LLWebRTCPeerConnectionInterface* peer_connection)
 {
-    std::vector<rtc::scoped_refptr<LLWebRTCPeerConnectionImpl>>::iterator it =
+    std::vector<webrtc::scoped_refptr<LLWebRTCPeerConnectionImpl>>::iterator it =
     std::find(mPeerConnections.begin(), mPeerConnections.end(), peer_connection);
     if (it != mPeerConnections.end())
     {
         mPeerConnections.erase(it);
-    }
-    if (mPeerConnections.empty())
-    {
-        setRecording(false);
-        setPlayout(false);
+        if (mPeerConnections.empty())
+        {
+            intSetMute(true);
+        }
     }
 }
 
@@ -729,7 +787,7 @@ void LLWebRTCPeerConnectionImpl::init(LLWebRTCImpl * webrtc_impl)
 }
 void LLWebRTCPeerConnectionImpl::terminate()
 {
-    mWebRTCImpl->SignalingBlockingCall(
+    mWebRTCImpl->PostSignalingTask(
         [this]()
         {
             if (mPeerConnection)
@@ -753,7 +811,6 @@ void LLWebRTCPeerConnectionImpl::terminate()
                         track->set_enabled(false);
                     }
                 }
-                mPeerConnection->SetAudioRecording(false);
 
                 mPeerConnection->Close();
                 if (mLocalStream)
@@ -840,7 +897,7 @@ bool LLWebRTCPeerConnectionImpl::initializeConnection(const LLWebRTCPeerConnecti
                 mDataChannel->RegisterObserver(this);
             }
 
-            cricket::AudioOptions audioOptions;
+            webrtc::AudioOptions audioOptions;
             audioOptions.auto_gain_control = true;
             audioOptions.echo_cancellation = true;
             audioOptions.noise_suppression = true;
@@ -848,7 +905,7 @@ bool LLWebRTCPeerConnectionImpl::initializeConnection(const LLWebRTCPeerConnecti
 
             mLocalStream = mPeerConnectionFactory->CreateLocalMediaStream("SLStream");
 
-            rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
+            webrtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
                 mPeerConnectionFactory->CreateAudioTrack("SLAudio", mPeerConnectionFactory->CreateAudioSource(audioOptions).get()));
             audio_track->set_enabled(false);
             mLocalStream->AddTrack(audio_track);
@@ -862,7 +919,7 @@ bool LLWebRTCPeerConnectionImpl::initializeConnection(const LLWebRTCPeerConnecti
                 webrtc::RtpParameters      params;
                 webrtc::RtpCodecParameters codecparam;
                 codecparam.name                       = "opus";
-                codecparam.kind                       = cricket::MEDIA_TYPE_AUDIO;
+                codecparam.kind                       = webrtc::MediaType::AUDIO;
                 codecparam.clock_rate                 = 48000;
                 codecparam.num_channels               = 2;
                 codecparam.parameters["stereo"]       = "1";
@@ -877,7 +934,7 @@ bool LLWebRTCPeerConnectionImpl::initializeConnection(const LLWebRTCPeerConnecti
                 webrtc::RtpParameters      params;
                 webrtc::RtpCodecParameters codecparam;
                 codecparam.name                       = "opus";
-                codecparam.kind                       = cricket::MEDIA_TYPE_AUDIO;
+                codecparam.kind                       = webrtc::MediaType::AUDIO;
                 codecparam.clock_rate                 = 48000;
                 codecparam.num_channels               = 2;
                 codecparam.parameters["stereo"]       = "1";
@@ -904,7 +961,6 @@ void LLWebRTCPeerConnectionImpl::enableSenderTracks(bool enable)
     // set_enabled shouldn't be done on the worker thread.
     if (mPeerConnection)
     {
-        mPeerConnection->SetAudioRecording(enable);
         auto senders = mPeerConnection->GetSenders();
         for (auto &sender : senders)
         {
@@ -938,7 +994,7 @@ void LLWebRTCPeerConnectionImpl::AnswerAvailable(const std::string &sdp)
                                    {
                                        RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->peer_connection_state();
                                        mPeerConnection->SetRemoteDescription(webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp),
-                                                                             rtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface>(this));
+                                                                             webrtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface>(this));
                                    }
                                });
 }
@@ -951,22 +1007,22 @@ void LLWebRTCPeerConnectionImpl::AnswerAvailable(const std::string &sdp)
 void LLWebRTCPeerConnectionImpl::setMute(bool mute)
 {
     EMicMuteState new_state = mute ? MUTE_MUTED : MUTE_UNMUTED;
-    if (new_state == mMute)
-    {
-        return; // no change
-    }
+
+    // even if mute hasn't changed, we still need to update the mute
+    // state on the connections to handle cases where the 'Default' device
+    // has changed in the OS (unplugged headset, etc.) which messes
+    // with the mute state.
+
     bool force_reset = mMute == MUTE_INITIAL && mute;
     bool enable = !mute;
     mMute = new_state;
 
+
     mWebRTCImpl->PostSignalingTask(
         [this, force_reset, enable]()
         {
         if (mPeerConnection)
         {
-            // SetAudioRecording must be called before enabling/disabling tracks.
-            mPeerConnection->SetAudioRecording(enable);
-
             auto senders = mPeerConnection->GetSenders();
 
             RTC_LOG(LS_INFO) << __FUNCTION__ << (mMute ? "disabling" : "enabling") << " streams count " << senders.size();
@@ -1046,14 +1102,14 @@ void LLWebRTCPeerConnectionImpl::setSendVolume(float volume)
 // PeerConnectionObserver implementation.
 //
 
-void LLWebRTCPeerConnectionImpl::OnAddTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface>                     receiver,
-                                            const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>> &streams)
+void LLWebRTCPeerConnectionImpl::OnAddTrack(webrtc::scoped_refptr<webrtc::RtpReceiverInterface>                     receiver,
+                                            const std::vector<webrtc::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.kind                       = webrtc::MediaType::AUDIO;
     codecparam.clock_rate                 = 48000;
     codecparam.num_channels               = 2;
     codecparam.parameters["stereo"]       = "1";
@@ -1062,12 +1118,12 @@ void LLWebRTCPeerConnectionImpl::OnAddTrack(rtc::scoped_refptr<webrtc::RtpReceiv
     receiver->SetParameters(params);
 }
 
-void LLWebRTCPeerConnectionImpl::OnRemoveTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver)
+void LLWebRTCPeerConnectionImpl::OnRemoveTrack(webrtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver)
 {
     RTC_LOG(LS_INFO) << __FUNCTION__ << " " << receiver->id();
 }
 
-void LLWebRTCPeerConnectionImpl::OnDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface> channel)
+void LLWebRTCPeerConnectionImpl::OnDataChannel(webrtc::scoped_refptr<webrtc::DataChannelInterface> channel)
 {
     if (mDataChannel)
     {
@@ -1154,23 +1210,23 @@ static std::string iceCandidateToTrickleString(const webrtc::IceCandidateInterfa
     candidate->candidate().address().ipaddr().ToString() << " " <<
     candidate->candidate().address().PortAsString() << " typ ";
 
-    if (candidate->candidate().type() == cricket::LOCAL_PORT_TYPE)
+    if (candidate->candidate().type() == webrtc::IceCandidateType::kHost)
     {
         candidate_stream << "host";
     }
-    else if (candidate->candidate().type() == cricket::STUN_PORT_TYPE)
+    else if (candidate->candidate().type() == webrtc::IceCandidateType::kSrflx)
     {
         candidate_stream << "srflx " <<
         "raddr " << candidate->candidate().related_address().ipaddr().ToString() << " " <<
         "rport " << candidate->candidate().related_address().PortAsString();
     }
-    else if (candidate->candidate().type() == cricket::RELAY_PORT_TYPE)
+    else if (candidate->candidate().type() == webrtc::IceCandidateType::kRelay)
     {
         candidate_stream << "relay " <<
         "raddr " << candidate->candidate().related_address().ipaddr().ToString() << " " <<
         "rport " << candidate->candidate().related_address().PortAsString();
     }
-    else if (candidate->candidate().type() == cricket::PRFLX_PORT_TYPE)
+    else if (candidate->candidate().type() == webrtc::IceCandidateType::kPrflx)
     {
         candidate_stream << "prflx " <<
         "raddr " << candidate->candidate().related_address().ipaddr().ToString() << " " <<
@@ -1265,7 +1321,7 @@ void LLWebRTCPeerConnectionImpl::OnSuccess(webrtc::SessionDescriptionInterface *
 
    mPeerConnection->SetLocalDescription(std::unique_ptr<webrtc::SessionDescriptionInterface>(
                                                      webrtc::CreateSessionDescription(webrtc::SdpType::kOffer, mangled_sdp)),
-                                                 rtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface>(this));
+                                                 webrtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface>(this));
 
 }
 
@@ -1375,7 +1431,7 @@ void LLWebRTCPeerConnectionImpl::sendData(const std::string& data, bool binary)
 {
     if (mDataChannel)
     {
-        rtc::CopyOnWriteBuffer cowBuffer(data.data(), data.length());
+        webrtc::CopyOnWriteBuffer cowBuffer(data.data(), data.length());
         webrtc::DataBuffer     buffer(cowBuffer, binary);
         mWebRTCImpl->PostNetworkTask([this, buffer]() {
                 if (mDataChannel)