Merging upstream version 126.0.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

20 files changed, 281 insertions, 197 deletions

diff --git a/dom/media/webrtc/MediaEnginePrefs.h b/dom/media/webrtc/MediaEnginePrefs.h
index cedb7f457c..de5daf0ad9 100644
--- a/dom/media/webrtc/MediaEnginePrefs.h
+++ b/dom/media/webrtc/MediaEnginePrefs.h
@@ -35,6 +35,7 @@ class MediaEnginePrefs {
         mNoiseOn(false),
         mTransientOn(false),
         mAgc2Forced(false),
+        mExpectDrift(-1),  // auto
         mAgc(0),
         mNoise(0),
         mChannels(0) {}
@@ -50,6 +51,7 @@ class MediaEnginePrefs {
   bool mNoiseOn;
   bool mTransientOn;
   bool mAgc2Forced;
+  int32_t mExpectDrift;
   int32_t mAgc;
   int32_t mNoise;
   int32_t mChannels;
diff --git a/dom/media/webrtc/MediaEngineWebRTCAudio.cpp b/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
index 9d778d411d..220dcf3bd8 100644
--- a/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
+++ b/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
@@ -20,6 +20,7 @@
 #include "mozilla/Sprintf.h"
 #include "mozilla/Logging.h"
 
+#include "api/audio/echo_canceller3_factory.h"
 #include "common_audio/include/audio_util.h"
 #include "modules/audio_processing/include/audio_processing.h"
 
@@ -146,22 +147,17 @@ nsresult MediaEngineWebRTCMicrophoneSource::Reconfigure(
   return NS_OK;
 }
 
-void MediaEngineWebRTCMicrophoneSource::ApplySettings(
+AudioProcessing::Config AudioInputProcessing::ConfigForPrefs(
     const MediaEnginePrefs& aPrefs) {
-  AssertIsOnOwningThread();
-
-  TRACE("ApplySettings");
-  MOZ_ASSERT(
-      mTrack,
-      "ApplySetting is to be called only after SetTrack has been called");
+  AudioProcessing::Config config;
 
-  mAudioProcessingConfig.pipeline.multi_channel_render = true;
-  mAudioProcessingConfig.pipeline.multi_channel_capture = true;
+  config.pipeline.multi_channel_render = true;
+  config.pipeline.multi_channel_capture = true;
 
-  mAudioProcessingConfig.echo_canceller.enabled = aPrefs.mAecOn;
-  mAudioProcessingConfig.echo_canceller.mobile_mode = aPrefs.mUseAecMobile;
+  config.echo_canceller.enabled = aPrefs.mAecOn;
+  config.echo_canceller.mobile_mode = aPrefs.mUseAecMobile;
 
-  if ((mAudioProcessingConfig.gain_controller1.enabled =
+  if ((config.gain_controller1.enabled =
            aPrefs.mAgcOn && !aPrefs.mAgc2Forced)) {
     auto mode = static_cast<AudioProcessing::Config::GainController1::Mode>(
         aPrefs.mAgc);
@@ -169,7 +165,7 @@ void MediaEngineWebRTCMicrophoneSource::ApplySettings(
         mode != AudioProcessing::Config::GainController1::kAdaptiveDigital &&
         mode != AudioProcessing::Config::GainController1::kFixedDigital) {
       LOG_ERROR("AudioInputProcessing %p Attempt to set invalid AGC mode %d",
-                mInputProcessing.get(), static_cast<int>(mode));
+                this, static_cast<int>(mode));
       mode = AudioProcessing::Config::GainController1::kAdaptiveDigital;
     }
 #if defined(WEBRTC_IOS) || defined(ATA) || defined(WEBRTC_ANDROID)
@@ -177,20 +173,20 @@ void MediaEngineWebRTCMicrophoneSource::ApplySettings(
       LOG_ERROR(
           "AudioInputProcessing %p Invalid AGC mode kAdaptiveAnalog on "
           "mobile",
-          mInputProcessing.get());
+          this);
       MOZ_ASSERT_UNREACHABLE(
           "Bad pref set in all.js or in about:config"
           " for the auto gain, on mobile.");
       mode = AudioProcessing::Config::GainController1::kFixedDigital;
     }
 #endif
-    mAudioProcessingConfig.gain_controller1.mode = mode;
+    config.gain_controller1.mode = mode;
   }
-  mAudioProcessingConfig.gain_controller2.enabled =
-      mAudioProcessingConfig.gain_controller2.adaptive_digital.enabled =
+  config.gain_controller2.enabled =
+      config.gain_controller2.adaptive_digital.enabled =
           aPrefs.mAgcOn && aPrefs.mAgc2Forced;
 
-  if ((mAudioProcessingConfig.noise_suppression.enabled = aPrefs.mNoiseOn)) {
+  if ((config.noise_suppression.enabled = aPrefs.mNoiseOn)) {
     auto level = static_cast<AudioProcessing::Config::NoiseSuppression::Level>(
         aPrefs.mNoise);
     if (level != AudioProcessing::Config::NoiseSuppression::kLow &&
@@ -200,49 +196,44 @@ void MediaEngineWebRTCMicrophoneSource::ApplySettings(
       LOG_ERROR(
           "AudioInputProcessing %p Attempt to set invalid noise suppression "
           "level %d",
-          mInputProcessing.get(), static_cast<int>(level));
+          this, static_cast<int>(level));
 
       level = AudioProcessing::Config::NoiseSuppression::kModerate;
     }
-    mAudioProcessingConfig.noise_suppression.level = level;
+    config.noise_suppression.level = level;
   }
 
-  mAudioProcessingConfig.transient_suppression.enabled = aPrefs.mTransientOn;
+  config.transient_suppression.enabled = aPrefs.mTransientOn;
+
+  config.high_pass_filter.enabled = aPrefs.mHPFOn;
 
-  mAudioProcessingConfig.high_pass_filter.enabled = aPrefs.mHPFOn;
+  return config;
+}
+
+void MediaEngineWebRTCMicrophoneSource::ApplySettings(
+    const MediaEnginePrefs& aPrefs) {
+  AssertIsOnOwningThread();
+
+  TRACE("ApplySettings");
+  MOZ_ASSERT(
+      mTrack,
+      "ApplySetting is to be called only after SetTrack has been called");
 
   RefPtr<MediaEngineWebRTCMicrophoneSource> that = this;
   CubebUtils::AudioDeviceID deviceID = mDeviceInfo->DeviceID();
   NS_DispatchToMainThread(NS_NewRunnableFunction(
-      __func__, [this, that, deviceID, track = mTrack, prefs = aPrefs,
-                 audioProcessingConfig = mAudioProcessingConfig] {
+      __func__, [this, that, deviceID, track = mTrack, prefs = aPrefs] {
         mSettings->mEchoCancellation.Value() = prefs.mAecOn;
         mSettings->mAutoGainControl.Value() = prefs.mAgcOn;
         mSettings->mNoiseSuppression.Value() = prefs.mNoiseOn;
         mSettings->mChannelCount.Value() = prefs.mChannels;
 
-        // The high-pass filter is not taken into account when activating the
-        // pass through, since it's not controllable from content.
-        bool passThrough = !(prefs.mAecOn || prefs.mAgcOn || prefs.mNoiseOn);
-
         if (track->IsDestroyed()) {
           return;
         }
         track->QueueControlMessageWithNoShutdown(
-            [track, deviceID, inputProcessing = mInputProcessing,
-             audioProcessingConfig, passThrough,
-             requestedInputChannelCount = prefs.mChannels] {
-              inputProcessing->ApplyConfig(track->Graph(),
-                                           audioProcessingConfig);
-              {
-                TRACE("SetRequestedInputChannelCount");
-                inputProcessing->SetRequestedInputChannelCount(
-                    track->Graph(), deviceID, requestedInputChannelCount);
-              }
-              {
-                TRACE("SetPassThrough");
-                inputProcessing->SetPassThrough(track->Graph(), passThrough);
-              }
+            [track, deviceID, prefs, inputProcessing = mInputProcessing] {
+              inputProcessing->ApplySettings(track->Graph(), deviceID, prefs);
             });
       }));
 }
@@ -408,57 +399,51 @@ void MediaEngineWebRTCMicrophoneSource::GetSettings(
 }
 
 AudioInputProcessing::AudioInputProcessing(uint32_t aMaxChannelCount)
-    : mAudioProcessing(AudioProcessingBuilder().Create().release()),
-      mRequestedInputChannelCount(aMaxChannelCount),
-      mSkipProcessing(false),
-      mInputDownmixBuffer(MAX_SAMPLING_FREQ * MAX_CHANNELS / 100),
+    : mInputDownmixBuffer(MAX_SAMPLING_FREQ * MAX_CHANNELS / 100),
       mEnabled(false),
       mEnded(false),
-      mPacketCount(0) {}
+      mPacketCount(0) {
+  mSettings.mChannels = static_cast<int32_t>(std::min<uint32_t>(
+      std::numeric_limits<int32_t>::max(), aMaxChannelCount));
+}
 
 void AudioInputProcessing::Disconnect(MediaTrackGraph* aGraph) {
   // This method is just for asserts.
   aGraph->AssertOnGraphThread();
 }
 
-bool AudioInputProcessing::PassThrough(MediaTrackGraph* aGraph) const {
+bool AudioInputProcessing::IsPassThrough(MediaTrackGraph* aGraph) const {
   aGraph->AssertOnGraphThread();
-  return mSkipProcessing;
+  // The high-pass filter is not taken into account when activating the
+  // pass through, since it's not controllable from content.
+  return !(mSettings.mAecOn || mSettings.mAgcOn || mSettings.mNoiseOn);
 }
 
-void AudioInputProcessing::SetPassThrough(MediaTrackGraph* aGraph,
-                                          bool aPassThrough) {
+void AudioInputProcessing::PassThroughChanged(MediaTrackGraph* aGraph) {
   aGraph->AssertOnGraphThread();
 
-  if (aPassThrough == mSkipProcessing) {
-    return;
-  }
-  mSkipProcessing = aPassThrough;
-
   if (!mEnabled) {
     MOZ_ASSERT(!mPacketizerInput);
     return;
   }
 
-  if (aPassThrough) {
-    // Turn on pass-through
+  if (IsPassThrough(aGraph)) {
+    // Switching to pass-through.  Clear state so that it doesn't affect any
+    // future processing, if re-enabled.
     ResetAudioProcessing(aGraph);
   } else {
-    // Turn off pass-through
+    // No longer pass-through.  Processing will not use old state.
+    // Packetizer setup is deferred until needed.
     MOZ_ASSERT(!mPacketizerInput);
-    EnsureAudioProcessing(aGraph, mRequestedInputChannelCount);
   }
 }
 
 uint32_t AudioInputProcessing::GetRequestedInputChannelCount() {
-  return mRequestedInputChannelCount;
+  return mSettings.mChannels;
 }
 
-void AudioInputProcessing::SetRequestedInputChannelCount(
-    MediaTrackGraph* aGraph, CubebUtils::AudioDeviceID aDeviceId,
-    uint32_t aRequestedInputChannelCount) {
-  mRequestedInputChannelCount = aRequestedInputChannelCount;
-
+void AudioInputProcessing::RequestedInputChannelCountChanged(
+    MediaTrackGraph* aGraph, CubebUtils::AudioDeviceID aDeviceId) {
   aGraph->ReevaluateInputDevice(aDeviceId);
 }
 
@@ -470,12 +455,7 @@ void AudioInputProcessing::Start(MediaTrackGraph* aGraph) {
   }
   mEnabled = true;
 
-  if (mSkipProcessing) {
-    return;
-  }
-
   MOZ_ASSERT(!mPacketizerInput);
-  EnsureAudioProcessing(aGraph, mRequestedInputChannelCount);
 }
 
 void AudioInputProcessing::Stop(MediaTrackGraph* aGraph) {
@@ -487,7 +467,7 @@ void AudioInputProcessing::Stop(MediaTrackGraph* aGraph) {
 
   mEnabled = false;
 
-  if (mSkipProcessing) {
+  if (IsPassThrough(aGraph)) {
     return;
   }
 
@@ -605,10 +585,11 @@ void AudioInputProcessing::Stop(MediaTrackGraph* aGraph) {
 //
 // The D(N) frames of data are just forwarded from input to output without any
 // processing
-void AudioInputProcessing::Process(MediaTrackGraph* aGraph, GraphTime aFrom,
-                                   GraphTime aTo, AudioSegment* aInput,
+void AudioInputProcessing::Process(AudioProcessingTrack* aTrack,
+                                   GraphTime aFrom, GraphTime aTo,
+                                   AudioSegment* aInput,
                                    AudioSegment* aOutput) {
-  aGraph->AssertOnGraphThread();
+  aTrack->AssertOnGraphThread();
   MOZ_ASSERT(aFrom <= aTo);
   MOZ_ASSERT(!mEnded);
 
@@ -617,10 +598,11 @@ void AudioInputProcessing::Process(MediaTrackGraph* aGraph, GraphTime aFrom,
     return;
   }
 
+  MediaTrackGraph* graph = aTrack->Graph();
   if (!mEnabled) {
     LOG_FRAME("(Graph %p, Driver %p) AudioInputProcessing %p Filling %" PRId64
               " frames of silence to output (disabled)",
-              aGraph, aGraph->CurrentDriver(), this, need);
+              graph, graph->CurrentDriver(), this, need);
     aOutput->AppendNullData(need);
     return;
   }
@@ -628,22 +610,20 @@ void AudioInputProcessing::Process(MediaTrackGraph* aGraph, GraphTime aFrom,
   MOZ_ASSERT(aInput->GetDuration() == need,
              "Wrong data length from input port source");
 
-  if (PassThrough(aGraph)) {
+  if (IsPassThrough(graph)) {
     LOG_FRAME(
         "(Graph %p, Driver %p) AudioInputProcessing %p Forwarding %" PRId64
         " frames of input data to output directly (PassThrough)",
-        aGraph, aGraph->CurrentDriver(), this, aInput->GetDuration());
+        graph, graph->CurrentDriver(), this, aInput->GetDuration());
     aOutput->AppendSegment(aInput);
     return;
   }
 
-  // SetPassThrough(false) must be called before reaching here.
-  MOZ_ASSERT(mPacketizerInput);
-  // If mRequestedInputChannelCount is updated, create a new packetizer. No
-  // need to change the pre-buffering since the rate is always the same. The
-  // frames left in the packetizer would be replaced by null data and then
-  // transferred to mSegment.
-  EnsureAudioProcessing(aGraph, mRequestedInputChannelCount);
+  // If the requested input channel count is updated, create a new
+  // packetizer. No need to change the pre-buffering since the rate is always
+  // the same. The frames left in the packetizer would be replaced by null
+  // data and then transferred to mSegment.
+  EnsurePacketizer(aTrack);
 
   // Preconditions of the audio-processing logic.
   MOZ_ASSERT(static_cast<uint32_t>(mSegment.GetDuration()) +
@@ -655,10 +635,10 @@ void AudioInputProcessing::Process(MediaTrackGraph* aGraph, GraphTime aFrom,
   MOZ_ASSERT(mSegment.GetDuration() >= 1);
   MOZ_ASSERT(mSegment.GetDuration() <= mPacketizerInput->mPacketSize);
 
-  PacketizeAndProcess(aGraph, *aInput);
+  PacketizeAndProcess(aTrack, *aInput);
   LOG_FRAME("(Graph %p, Driver %p) AudioInputProcessing %p Buffer has %" PRId64
             " frames of data now, after packetizing and processing",
-            aGraph, aGraph->CurrentDriver(), this, mSegment.GetDuration());
+            graph, graph->CurrentDriver(), this, mSegment.GetDuration());
 
   // By setting pre-buffering to the number of frames of one packet, and
   // because the maximum number of frames stuck in the packetizer before
@@ -669,8 +649,7 @@ void AudioInputProcessing::Process(MediaTrackGraph* aGraph, GraphTime aFrom,
   mSegment.RemoveLeading(need);
   LOG_FRAME("(Graph %p, Driver %p) AudioInputProcessing %p moving %" PRId64
             " frames of data to output, leaving %" PRId64 " frames in buffer",
-            aGraph, aGraph->CurrentDriver(), this, need,
-            mSegment.GetDuration());
+            graph, graph->CurrentDriver(), this, need, mSegment.GetDuration());
 
   // Postconditions of the audio-processing logic.
   MOZ_ASSERT(static_cast<uint32_t>(mSegment.GetDuration()) +
@@ -680,16 +659,16 @@ void AudioInputProcessing::Process(MediaTrackGraph* aGraph, GraphTime aFrom,
   MOZ_ASSERT(mSegment.GetDuration() <= mPacketizerInput->mPacketSize);
 }
 
-void AudioInputProcessing::ProcessOutputData(MediaTrackGraph* aGraph,
+void AudioInputProcessing::ProcessOutputData(AudioProcessingTrack* aTrack,
                                              const AudioChunk& aChunk) {
   MOZ_ASSERT(aChunk.ChannelCount() > 0);
-  aGraph->AssertOnGraphThread();
+  aTrack->AssertOnGraphThread();
 
-  if (!mEnabled || PassThrough(aGraph)) {
+  if (!mEnabled || IsPassThrough(aTrack->Graph())) {
     return;
   }
 
-  TrackRate sampleRate = aGraph->GraphRate();
+  TrackRate sampleRate = aTrack->mSampleRate;
   uint32_t framesPerPacket = GetPacketSize(sampleRate);  // in frames
   // Downmix from aChannels to MAX_CHANNELS if needed.
   uint32_t channelCount =
@@ -727,6 +706,7 @@ void AudioInputProcessing::ProcessOutputData(MediaTrackGraph* aGraph,
 
     if (mOutputBufferFrameCount == framesPerPacket) {
       // Have a complete packet.  Analyze it.
+      EnsureAudioProcessing(aTrack);
       for (uint32_t channel = 0; channel < channelCount; channel++) {
         channelPtrs[channel] = &mOutputBuffer[channel * framesPerPacket];
       }
@@ -743,14 +723,15 @@ void AudioInputProcessing::ProcessOutputData(MediaTrackGraph* aGraph,
 }
 
 // Only called if we're not in passthrough mode
-void AudioInputProcessing::PacketizeAndProcess(MediaTrackGraph* aGraph,
+void AudioInputProcessing::PacketizeAndProcess(AudioProcessingTrack* aTrack,
                                                const AudioSegment& aSegment) {
-  MOZ_ASSERT(!PassThrough(aGraph),
+  MediaTrackGraph* graph = aTrack->Graph();
+  MOZ_ASSERT(!IsPassThrough(graph),
              "This should be bypassed when in PassThrough mode.");
   MOZ_ASSERT(mEnabled);
   MOZ_ASSERT(mPacketizerInput);
   MOZ_ASSERT(mPacketizerInput->mPacketSize ==
-             GetPacketSize(aGraph->GraphRate()));
+             GetPacketSize(aTrack->mSampleRate));
 
   // Calculate number of the pending frames in mChunksInPacketizer.
   auto pendingFrames = [&]() {
@@ -792,7 +773,7 @@ void AudioInputProcessing::PacketizeAndProcess(MediaTrackGraph* aGraph,
   LOG_FRAME(
       "(Graph %p, Driver %p) AudioInputProcessing %p Packetizing %zu frames. "
       "Packetizer has %u frames (enough for %u packets) now",
-      aGraph, aGraph->CurrentDriver(), this, frameCount,
+      graph, graph->CurrentDriver(), this, frameCount,
       mPacketizerInput->FramesAvailable(),
       mPacketizerInput->PacketsAvailable());
 
@@ -850,9 +831,10 @@ void AudioInputProcessing::PacketizeAndProcess(MediaTrackGraph* aGraph,
                    deinterleavedPacketizedInputDataChannelPointers.Elements());
     }
 
-    StreamConfig inputConfig(aGraph->GraphRate(), channelCountInput);
+    StreamConfig inputConfig(aTrack->mSampleRate, channelCountInput);
     StreamConfig outputConfig = inputConfig;
 
+    EnsureAudioProcessing(aTrack);
     // Bug 1404965: Get the right delay here, it saves some work down the line.
     mAudioProcessing->set_stream_delay_ms(0);
 
@@ -958,7 +940,7 @@ void AudioInputProcessing::PacketizeAndProcess(MediaTrackGraph* aGraph,
         "(Graph %p, Driver %p) AudioInputProcessing %p Appending %u frames of "
         "packetized audio, leaving %u frames in packetizer (%" PRId64
         " frames in mChunksInPacketizer)",
-        aGraph, aGraph->CurrentDriver(), this, mPacketizerInput->mPacketSize,
+        graph, graph->CurrentDriver(), this, mPacketizerInput->mPacketSize,
         mPacketizerInput->FramesAvailable(), pendingFrames());
 
     // Postcondition of the Principal-labelling logic.
@@ -971,17 +953,36 @@ void AudioInputProcessing::DeviceChanged(MediaTrackGraph* aGraph) {
   aGraph->AssertOnGraphThread();
 
   // Reset some processing
-  mAudioProcessing->Initialize();
+  if (mAudioProcessing) {
+    mAudioProcessing->Initialize();
+  }
   LOG_FRAME(
       "(Graph %p, Driver %p) AudioInputProcessing %p Reinitializing audio "
       "processing",
       aGraph, aGraph->CurrentDriver(), this);
 }
 
-void AudioInputProcessing::ApplyConfig(MediaTrackGraph* aGraph,
-                                       const AudioProcessing::Config& aConfig) {
+void AudioInputProcessing::ApplySettings(MediaTrackGraph* aGraph,
+                                         CubebUtils::AudioDeviceID aDeviceID,
+                                         const MediaEnginePrefs& aSettings) {
+  TRACE("AudioInputProcessing::ApplySettings");
   aGraph->AssertOnGraphThread();
-  mAudioProcessing->ApplyConfig(aConfig);
+
+  // Read previous state from mSettings.
+  uint32_t oldChannelCount = GetRequestedInputChannelCount();
+  bool wasPassThrough = IsPassThrough(aGraph);
+
+  mSettings = aSettings;
+  if (mAudioProcessing) {
+    mAudioProcessing->ApplyConfig(ConfigForPrefs(aSettings));
+  }
+
+  if (oldChannelCount != GetRequestedInputChannelCount()) {
+    RequestedInputChannelCountChanged(aGraph, aDeviceID);
+  }
+  if (wasPassThrough != IsPassThrough(aGraph)) {
+    PassThroughChanged(aGraph);
+  }
 }
 
 void AudioInputProcessing::End() {
@@ -995,14 +996,15 @@ TrackTime AudioInputProcessing::NumBufferedFrames(
   return mSegment.GetDuration();
 }
 
-void AudioInputProcessing::EnsureAudioProcessing(MediaTrackGraph* aGraph,
-                                                 uint32_t aChannels) {
-  aGraph->AssertOnGraphThread();
-  MOZ_ASSERT(aChannels > 0);
+void AudioInputProcessing::EnsurePacketizer(AudioProcessingTrack* aTrack) {
+  aTrack->AssertOnGraphThread();
   MOZ_ASSERT(mEnabled);
-  MOZ_ASSERT(!mSkipProcessing);
+  MediaTrackGraph* graph = aTrack->Graph();
+  MOZ_ASSERT(!IsPassThrough(graph));
 
-  if (mPacketizerInput && mPacketizerInput->mChannels == aChannels) {
+  uint32_t channelCount = GetRequestedInputChannelCount();
+  MOZ_ASSERT(channelCount > 0);
+  if (mPacketizerInput && mPacketizerInput->mChannels == channelCount) {
     return;
   }
 
@@ -1010,7 +1012,7 @@ void AudioInputProcessing::EnsureAudioProcessing(MediaTrackGraph* aGraph,
   // need to change pre-buffering since the packet size is the same as the old
   // one, since the rate is a constant.
   MOZ_ASSERT_IF(mPacketizerInput, mPacketizerInput->mPacketSize ==
-                                      GetPacketSize(aGraph->GraphRate()));
+                                      GetPacketSize(aTrack->mSampleRate));
   bool needPreBuffering = !mPacketizerInput;
   if (mPacketizerInput) {
     const TrackTime numBufferedFrames =
@@ -1020,24 +1022,62 @@ void AudioInputProcessing::EnsureAudioProcessing(MediaTrackGraph* aGraph,
     mChunksInPacketizer.clear();
   }
 
-  mPacketizerInput.emplace(GetPacketSize(aGraph->GraphRate()), aChannels);
+  mPacketizerInput.emplace(GetPacketSize(aTrack->mSampleRate), channelCount);
 
   if (needPreBuffering) {
     LOG_FRAME(
         "(Graph %p, Driver %p) AudioInputProcessing %p: Adding %u frames of "
         "silence as pre-buffering",
-        aGraph, aGraph->CurrentDriver(), this, mPacketizerInput->mPacketSize);
+        graph, graph->CurrentDriver(), this, mPacketizerInput->mPacketSize);
 
     AudioSegment buffering;
     buffering.AppendNullData(
         static_cast<TrackTime>(mPacketizerInput->mPacketSize));
-    PacketizeAndProcess(aGraph, buffering);
+    PacketizeAndProcess(aTrack, buffering);
+  }
+}
+
+void AudioInputProcessing::EnsureAudioProcessing(AudioProcessingTrack* aTrack) {
+  aTrack->AssertOnGraphThread();
+
+  MediaTrackGraph* graph = aTrack->Graph();
+  // If the AEC might need to deal with drift then inform it of this and it
+  // will be less conservative about echo suppression.  This can lead to some
+  // suppression of non-echo signal, so do this only when drift is expected.
+  // https://bugs.chromium.org/p/webrtc/issues/detail?id=11985#c2
+  bool haveAECAndDrift = mSettings.mAecOn;
+  if (haveAECAndDrift) {
+    if (mSettings.mExpectDrift < 0) {
+      haveAECAndDrift =
+          graph->OutputForAECMightDrift() ||
+          aTrack->GetDeviceInputTrackGraphThread()->AsNonNativeInputTrack();
+    } else {
+      haveAECAndDrift = mSettings.mExpectDrift > 0;
+    }
+  }
+  if (!mAudioProcessing || haveAECAndDrift != mHadAECAndDrift) {
+    TRACE("AudioProcessing creation");
+    LOG("Track %p AudioInputProcessing %p creating AudioProcessing. "
+        "aec+drift: %s",
+        aTrack, this, haveAECAndDrift ? "Y" : "N");
+    mHadAECAndDrift = haveAECAndDrift;
+    AudioProcessingBuilder builder;
+    builder.SetConfig(ConfigForPrefs(mSettings));
+    if (haveAECAndDrift) {
+      // Setting an EchoControlFactory always enables AEC, overriding
+      // Config::echo_canceller.enabled, so do this only when AEC is enabled.
+      EchoCanceller3Config aec3Config;
+      aec3Config.echo_removal_control.has_clock_drift = true;
+      builder.SetEchoControlFactory(
+          std::make_unique<EchoCanceller3Factory>(aec3Config));
+    }
+    mAudioProcessing.reset(builder.Create().release());
   }
 }
 
 void AudioInputProcessing::ResetAudioProcessing(MediaTrackGraph* aGraph) {
   aGraph->AssertOnGraphThread();
-  MOZ_ASSERT(mSkipProcessing || !mEnabled);
+  MOZ_ASSERT(IsPassThrough(aGraph) || !mEnabled);
   MOZ_ASSERT(mPacketizerInput);
 
   LOG_FRAME(
@@ -1047,7 +1087,9 @@ void AudioInputProcessing::ResetAudioProcessing(MediaTrackGraph* aGraph) {
 
   // Reset AudioProcessing so that if we resume processing in the future it
   // doesn't depend on old state.
-  mAudioProcessing->Initialize();
+  if (mAudioProcessing) {
+    mAudioProcessing->Initialize();
+  }
 
   MOZ_ASSERT(static_cast<uint32_t>(mSegment.GetDuration()) +
                  mPacketizerInput->FramesAvailable() ==
@@ -1124,9 +1166,8 @@ void AudioProcessingTrack::ProcessInput(GraphTime aFrom, GraphTime aTo,
     } else {
       MOZ_ASSERT(mInputs.Length() == 1);
       AudioSegment data;
-      DeviceInputConsumerTrack::GetInputSourceData(data, mInputs[0], aFrom,
-                                                   aTo);
-      mInputProcessing->Process(Graph(), aFrom, aTo, &data,
+      DeviceInputConsumerTrack::GetInputSourceData(data, aFrom, aTo);
+      mInputProcessing->Process(this, aFrom, aTo, &data,
                                 GetData<AudioSegment>());
     }
     MOZ_ASSERT(TrackTimeToGraphTime(GetEnd()) == aTo);
@@ -1142,7 +1183,7 @@ void AudioProcessingTrack::NotifyOutputData(MediaTrackGraph* aGraph,
   MOZ_ASSERT(mGraph == aGraph, "Cannot feed audio output to another graph");
   AssertOnGraphThread();
   if (mInputProcessing) {
-    mInputProcessing->ProcessOutputData(aGraph, aChunk);
+    mInputProcessing->ProcessOutputData(this, aChunk);
   }
 }
 
diff --git a/dom/media/webrtc/MediaEngineWebRTCAudio.h b/dom/media/webrtc/MediaEngineWebRTCAudio.h
index e71b5ef826..6b1fbf0089 100644
--- a/dom/media/webrtc/MediaEngineWebRTCAudio.h
+++ b/dom/media/webrtc/MediaEngineWebRTCAudio.h
@@ -91,8 +91,7 @@ class MediaEngineWebRTCMicrophoneSource : public MediaEngineSource {
   // Current state of the resource for this source.
   MediaEngineSourceState mState;
 
-  // The current preferences that will be forwarded to mAudioProcessingConfig
-  // below.
+  // The current preferences that will be forwarded to mInputProcessing below.
   MediaEnginePrefs mCurrentPrefs;
 
   // The AudioProcessingTrack used to inteface with the MediaTrackGraph. Set in
@@ -101,10 +100,6 @@ class MediaEngineWebRTCMicrophoneSource : public MediaEngineSource {
 
   // See note at the top of this class.
   RefPtr<AudioInputProcessing> mInputProcessing;
-
-  // Copy of the config currently applied to AudioProcessing through
-  // mInputProcessing.
-  webrtc::AudioProcessing::Config mAudioProcessingConfig;
 };
 
 // This class is created on the MediaManager thread, and then exclusively used
@@ -113,15 +108,16 @@ class MediaEngineWebRTCMicrophoneSource : public MediaEngineSource {
 class AudioInputProcessing : public AudioDataListener {
  public:
   explicit AudioInputProcessing(uint32_t aMaxChannelCount);
-  void Process(MediaTrackGraph* aGraph, GraphTime aFrom, GraphTime aTo,
+  void Process(AudioProcessingTrack* aTrack, GraphTime aFrom, GraphTime aTo,
                AudioSegment* aInput, AudioSegment* aOutput);
 
-  void ProcessOutputData(MediaTrackGraph* aGraph, const AudioChunk& aChunk);
+  void ProcessOutputData(AudioProcessingTrack* aTrack,
+                         const AudioChunk& aChunk);
   bool IsVoiceInput(MediaTrackGraph* aGraph) const override {
     // If we're passing data directly without AEC or any other process, this
     // means that all voice-processing has been disabled intentionaly. In this
     // case, consider that the device is not used for voice input.
-    return !PassThrough(aGraph);
+    return !IsPassThrough(aGraph);
   }
 
   void Start(MediaTrackGraph* aGraph);
@@ -135,23 +131,20 @@ class AudioInputProcessing : public AudioDataListener {
 
   void Disconnect(MediaTrackGraph* aGraph) override;
 
-  void PacketizeAndProcess(MediaTrackGraph* aGraph,
+  void PacketizeAndProcess(AudioProcessingTrack* aTrack,
                            const AudioSegment& aSegment);
 
-  void SetPassThrough(MediaTrackGraph* aGraph, bool aPassThrough);
   uint32_t GetRequestedInputChannelCount();
-  void SetRequestedInputChannelCount(MediaTrackGraph* aGraph,
-                                     CubebUtils::AudioDeviceID aDeviceId,
-                                     uint32_t aRequestedInputChannelCount);
-  // This is true when all processing is disabled, we can skip
+  // This is true when all processing is disabled, in which case we can skip
   // packetization, resampling and other processing passes.
-  bool PassThrough(MediaTrackGraph* aGraph) const;
+  bool IsPassThrough(MediaTrackGraph* aGraph) const;
 
   // This allow changing the APM options, enabling or disabling processing
-  // steps. The config gets applied the next time we're about to process input
+  // steps. The settings get applied the next time we're about to process input
   // data.
-  void ApplyConfig(MediaTrackGraph* aGraph,
-                   const webrtc::AudioProcessing::Config& aConfig);
+  void ApplySettings(MediaTrackGraph* aGraph,
+                     CubebUtils::AudioDeviceID aDeviceID,
+                     const MediaEnginePrefs& aSettings);
 
   void End();
 
@@ -164,9 +157,18 @@ class AudioInputProcessing : public AudioDataListener {
 
   bool IsEnded() const { return mEnded; }
 
+  // For testing:
+  bool HadAECAndDrift() const { return mHadAECAndDrift; }
+
  private:
   ~AudioInputProcessing() = default;
-  void EnsureAudioProcessing(MediaTrackGraph* aGraph, uint32_t aChannels);
+  webrtc::AudioProcessing::Config ConfigForPrefs(
+      const MediaEnginePrefs& aPrefs);
+  void PassThroughChanged(MediaTrackGraph* aGraph);
+  void RequestedInputChannelCountChanged(MediaTrackGraph* aGraph,
+                                         CubebUtils::AudioDeviceID aDeviceId);
+  void EnsurePacketizer(AudioProcessingTrack* aTrack);
+  void EnsureAudioProcessing(AudioProcessingTrack* aTrack);
   void ResetAudioProcessing(MediaTrackGraph* aGraph);
   PrincipalHandle GetCheckedPrincipal(const AudioSegment& aSegment);
   // This implements the processing algoritm to apply to the input (e.g. a
@@ -174,17 +176,16 @@ class AudioInputProcessing : public AudioDataListener {
   // class only accepts audio chunks of 10ms. It has two inputs and one output:
   // it is fed the speaker data and the microphone data. It outputs processed
   // input data.
-  const UniquePtr<webrtc::AudioProcessing> mAudioProcessing;
+  UniquePtr<webrtc::AudioProcessing> mAudioProcessing;
+  // Whether mAudioProcessing was created for AEC with clock drift.
+  // Meaningful only when mAudioProcessing is non-null;
+  bool mHadAECAndDrift = false;
   // Packetizer to be able to feed 10ms packets to the input side of
   // mAudioProcessing. Not used if the processing is bypassed.
   Maybe<AudioPacketizer<AudioDataValue, float>> mPacketizerInput;
-  // The number of channels asked for by content, after clamping to the range of
-  // legal channel count for this particular device.
-  uint32_t mRequestedInputChannelCount;
-  // mSkipProcessing is true if none of the processing passes are enabled,
-  // because of prefs or constraints. This allows simply copying the audio into
-  // the MTG, skipping resampling and the whole webrtc.org code.
-  bool mSkipProcessing;
+  // The current settings from about:config preferences and content-provided
+  // constraints.
+  MediaEnginePrefs mSettings;
   // Buffer for up to one 10ms packet of planar mixed audio output for the
   // reverse-stream (speaker data) of mAudioProcessing AEC.
   // Length is packet size * channel count, regardless of how many frames are
diff --git a/dom/media/webrtc/jsapi/PeerConnectionCtx.cpp b/dom/media/webrtc/jsapi/PeerConnectionCtx.cpp
index d293fa0be6..b9b9ab8fc5 100644
--- a/dom/media/webrtc/jsapi/PeerConnectionCtx.cpp
+++ b/dom/media/webrtc/jsapi/PeerConnectionCtx.cpp
@@ -129,11 +129,12 @@ class DummyAudioProcessing : public AudioProcessing {
   }
   void set_stream_key_pressed(bool) override { MOZ_CRASH("Unexpected call"); }
   bool CreateAndAttachAecDump(absl::string_view, int64_t,
-                              rtc::TaskQueue*) override {
+                              absl::Nonnull<TaskQueueBase*>) override {
     MOZ_CRASH("Unexpected call");
     return false;
   }
-  bool CreateAndAttachAecDump(FILE*, int64_t, rtc::TaskQueue*) override {
+  bool CreateAndAttachAecDump(FILE*, int64_t,
+                              absl::Nonnull<TaskQueueBase*>) override {
     MOZ_CRASH("Unexpected call");
     return false;
   }
diff --git a/dom/media/webrtc/jsapi/RTCRtpScriptTransform.cpp b/dom/media/webrtc/jsapi/RTCRtpScriptTransform.cpp
index 43f34c456f..8fa0bade00 100644
--- a/dom/media/webrtc/jsapi/RTCRtpScriptTransform.cpp
+++ b/dom/media/webrtc/jsapi/RTCRtpScriptTransform.cpp
@@ -47,6 +47,22 @@ already_AddRefed<RTCRtpScriptTransform> RTCRtpScriptTransform::Constructor(
     aRv.Throw(NS_ERROR_FAILURE);
     return nullptr;
   }
+
+  // The spec currently fails to describe what to do when the worker is closing
+  // or closed; the following placeholder text can be found in the spec at:
+  // https://w3c.github.io/webrtc-encoded-transform/#dom-rtcrtpscripttransform-rtcrtpscripttransform
+  //
+  // > FIXME: Describe error handling (worker closing flag true at
+  // > RTCRtpScriptTransform creation time. And worker being terminated while
+  // > transform is processing data).
+  //
+  // Because our worker runnables do not like to be pointed at a nonexistant
+  // worker, we throw in this case.
+  if (!aWorker.IsEligibleForMessaging()) {
+    aRv.Throw(NS_ERROR_FAILURE);
+    return nullptr;
+  }
+
   auto newTransform = MakeRefPtr<RTCRtpScriptTransform>(ownerWindow);
   RefPtr<RTCTransformEventRunnable> runnable =
       new RTCTransformEventRunnable(aWorker, &newTransform->GetProxy());
diff --git a/dom/media/webrtc/jsapi/RTCRtpScriptTransformer.cpp b/dom/media/webrtc/jsapi/RTCRtpScriptTransformer.cpp
index 126020a94f..f2fbd6d637 100644
--- a/dom/media/webrtc/jsapi/RTCRtpScriptTransformer.cpp
+++ b/dom/media/webrtc/jsapi/RTCRtpScriptTransformer.cpp
@@ -148,7 +148,7 @@ WritableStreamRTCFrameSink::WritableStreamRTCFrameSink(
 
 WritableStreamRTCFrameSink::~WritableStreamRTCFrameSink() = default;
 
-already_AddRefed<Promise> WritableStreamRTCFrameSink::WriteCallback(
+already_AddRefed<Promise> WritableStreamRTCFrameSink::WriteCallbackImpl(
     JSContext* aCx, JS::Handle<JS::Value> aChunk,
     WritableStreamDefaultController& aController, ErrorResult& aError) {
   // Spec does not say to do this right now. Might be a spec bug, needs
diff --git a/dom/media/webrtc/jsapi/RTCRtpScriptTransformer.h b/dom/media/webrtc/jsapi/RTCRtpScriptTransformer.h
index 6d61ac3cd5..7a22612254 100644
--- a/dom/media/webrtc/jsapi/RTCRtpScriptTransformer.h
+++ b/dom/media/webrtc/jsapi/RTCRtpScriptTransformer.h
@@ -87,7 +87,7 @@ class WritableStreamRTCFrameSink final
   NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(WritableStreamRTCFrameSink,
                                            UnderlyingSinkAlgorithmsWrapper)
 
-  already_AddRefed<Promise> WriteCallback(
+  already_AddRefed<Promise> WriteCallbackImpl(
       JSContext* aCx, JS::Handle<JS::Value> aChunk,
       WritableStreamDefaultController& aController,
       ErrorResult& aError) override;
diff --git a/dom/media/webrtc/jsapi/WebrtcGlobalInformation.cpp b/dom/media/webrtc/jsapi/WebrtcGlobalInformation.cpp
index 4e4bf9ab93..eabf7ee335 100644
--- a/dom/media/webrtc/jsapi/WebrtcGlobalInformation.cpp
+++ b/dom/media/webrtc/jsapi/WebrtcGlobalInformation.cpp
@@ -234,7 +234,7 @@ void WebrtcGlobalInformation::GetStatsHistorySince(
   auto statsAfter = aAfter.WasPassed() ? Some(aAfter.Value()) : Nothing();
   auto sdpAfter = aSdpAfter.WasPassed() ? Some(aSdpAfter.Value()) : Nothing();
 
-  WebrtcGlobalStatsHistory::GetHistory(pcIdFilter).apply([&](auto& hist) {
+  WebrtcGlobalStatsHistory::GetHistory(pcIdFilter).apply([&](const auto& hist) {
     if (!history.mReports.AppendElements(hist->Since(statsAfter), fallible)) {
       mozalloc_handle_oom(0);
     }
diff --git a/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp b/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp
index 49f049cd21..91ad0d848c 100644
--- a/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp
+++ b/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp
@@ -907,7 +907,7 @@ RtpExtList WebrtcAudioConduit::FilterExtensions(LocalDirection aDirection,
 
 webrtc::SdpAudioFormat WebrtcAudioConduit::CodecConfigToLibwebrtcFormat(
     const AudioCodecConfig& aConfig) {
-  webrtc::SdpAudioFormat::Parameters parameters;
+  webrtc::CodecParameterMap parameters;
   if (aConfig.mName == kOpusCodecName) {
     if (aConfig.mChannels == 2) {
       parameters[kCodecParamStereo] = kParamValueTrue;
diff --git a/dom/media/webrtc/libwebrtcglue/VideoConduit.cpp b/dom/media/webrtc/libwebrtcglue/VideoConduit.cpp
index 73e59f5ee2..5862237711 100644
--- a/dom/media/webrtc/libwebrtcglue/VideoConduit.cpp
+++ b/dom/media/webrtc/libwebrtcglue/VideoConduit.cpp
@@ -190,7 +190,7 @@ webrtc::VideoCodecType SupportedCodecType(webrtc::VideoCodecType aType) {
 rtc::scoped_refptr<webrtc::VideoEncoderConfig::EncoderSpecificSettings>
 ConfigureVideoEncoderSettings(const VideoCodecConfig& aConfig,
                               const WebrtcVideoConduit* aConduit,
-                              webrtc::SdpVideoFormat::Parameters& aParameters) {
+                              webrtc::CodecParameterMap& aParameters) {
   bool is_screencast =
       aConduit->CodecMode() == webrtc::VideoCodecMode::kScreensharing;
   // No automatic resizing when using simulcast or screencast.
diff --git a/dom/media/webrtc/libwebrtcglue/WebrtcCallWrapper.cpp b/dom/media/webrtc/libwebrtcglue/WebrtcCallWrapper.cpp
index 824f1cf6eb..b03c1772c4 100644
--- a/dom/media/webrtc/libwebrtcglue/WebrtcCallWrapper.cpp
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcCallWrapper.cpp
@@ -11,6 +11,8 @@
 #include "TaskQueueWrapper.h"
 
 // libwebrtc includes
+#include "api/environment/environment.h"
+#include "api/environment/environment_factory.h"
 #include "call/rtp_transport_controller_send_factory.h"
 
 namespace mozilla {
@@ -28,17 +30,16 @@ namespace mozilla {
       std::move(eventLog), std::move(taskQueueFactory), aTimestampMaker,
       std::move(aShutdownTicket));
 
+  webrtc::Environment env = CreateEnvironment(
+      wrapper->mEventLog.get(), wrapper->mClock.GetRealTimeClockRaw(),
+      wrapper->mTaskQueueFactory.get(), aSharedState->mTrials.get());
+
   wrapper->mCallThread->Dispatch(
-      NS_NewRunnableFunction(__func__, [wrapper, aSharedState] {
-        webrtc::CallConfig config(wrapper->mEventLog.get());
+      NS_NewRunnableFunction(__func__, [wrapper, aSharedState, env] {
+        webrtc::CallConfig config(env, nullptr);
         config.audio_state =
             webrtc::AudioState::Create(aSharedState->mAudioStateConfig);
-        config.task_queue_factory = wrapper->mTaskQueueFactory.get();
-        config.trials = aSharedState->mTrials.get();
-        wrapper->SetCall(WrapUnique(webrtc::Call::Create(
-            config, &wrapper->mClock,
-            webrtc::RtpTransportControllerSendFactory().Create(
-                config.ExtractTransportConfig(), &wrapper->mClock)).release()));
+        wrapper->SetCall(WrapUnique(webrtc::Call::Create(config).release()));
       }));
 
   return wrapper;
diff --git a/dom/media/webrtc/libwebrtcglue/WebrtcGmpVideoCodec.h b/dom/media/webrtc/libwebrtcglue/WebrtcGmpVideoCodec.h
index 865f9afff0..b8ee44c6b3 100644
--- a/dom/media/webrtc/libwebrtcglue/WebrtcGmpVideoCodec.h
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcGmpVideoCodec.h
@@ -289,7 +289,7 @@ class WebrtcGmpVideoEncoder : public GMPVideoEncoderCallbackProxy,
   GMPVideoHost* mHost;
   GMPVideoCodec mCodecParams;
   uint32_t mMaxPayloadSize;
-  const webrtc::SdpVideoFormat::Parameters mFormatParams;
+  const webrtc::CodecParameterMap mFormatParams;
   webrtc::CodecSpecificInfo mCodecSpecificInfo;
   webrtc::H264BitstreamParser mH264BitstreamParser;
   // Protects mCallback
diff --git a/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.cpp b/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.cpp
index 844542cd0d..f5240ffa22 100644
--- a/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.cpp
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.cpp
@@ -75,7 +75,7 @@ static const char* PacketModeStr(const webrtc::CodecSpecificInfo& aInfo) {
 }
 
 static std::pair<H264_PROFILE, H264_LEVEL> ConvertProfileLevel(
-    const webrtc::SdpVideoFormat::Parameters& aParameters) {
+    const webrtc::CodecParameterMap& aParameters) {
   const absl::optional<webrtc::H264ProfileLevelId> profileLevel =
       webrtc::ParseSdpForH264ProfileLevelId(aParameters);
 
@@ -143,9 +143,9 @@ WebrtcMediaDataEncoder::~WebrtcMediaDataEncoder() {
   }
 }
 
-static void InitCodecSpecficInfo(
-    webrtc::CodecSpecificInfo& aInfo, const webrtc::VideoCodec* aCodecSettings,
-    const webrtc::SdpVideoFormat::Parameters& aParameters) {
+static void InitCodecSpecficInfo(webrtc::CodecSpecificInfo& aInfo,
+                                 const webrtc::VideoCodec* aCodecSettings,
+                                 const webrtc::CodecParameterMap& aParameters) {
   MOZ_ASSERT(aCodecSettings);
 
   aInfo.codecType = aCodecSettings->codecType;
@@ -290,13 +290,11 @@ already_AddRefed<MediaDataEncoder> WebrtcMediaDataEncoder::CreateEncoder(
       MOZ_MAKE_COMPILER_ASSUME_IS_UNREACHABLE("Unsupported codec type");
   }
   EncoderConfig config(
-      type, {aCodecSettings->width, aCodecSettings->height},
-      MediaDataEncoder::Usage::Realtime, MediaDataEncoder::PixelFormat::YUV420P,
-      MediaDataEncoder::PixelFormat::YUV420P, aCodecSettings->maxFramerate,
-      keyframeInterval, mBitrateAdjuster.GetTargetBitrateBps(),
-      MediaDataEncoder::BitrateMode::Variable,
-      MediaDataEncoder::HardwarePreference::None,
-      MediaDataEncoder::ScalabilityMode::None, specific);
+      type, {aCodecSettings->width, aCodecSettings->height}, Usage::Realtime,
+      dom::ImageBitmapFormat::YUV420P, dom::ImageBitmapFormat::YUV420P,
+      aCodecSettings->maxFramerate, keyframeInterval,
+      mBitrateAdjuster.GetTargetBitrateBps(), BitrateMode::Variable,
+      HardwarePreference::None, ScalabilityMode::None, specific);
   return mFactory->CreateEncoder(config, mTaskQueue);
 }
 
diff --git a/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.h b/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.h
index 9d750e85b2..0c2070f6a9 100644
--- a/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.h
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.h
@@ -65,7 +65,7 @@ class WebrtcMediaDataEncoder : public RefCountedWebrtcVideoEncoder {
   MediaResult mError = NS_OK;
 
   VideoInfo mInfo;
-  webrtc::SdpVideoFormat::Parameters mFormatParams;
+  webrtc::CodecParameterMap mFormatParams;
   webrtc::CodecSpecificInfo mCodecSpecific;
   webrtc::BitrateAdjuster mBitrateAdjuster;
   uint32_t mMaxFrameRate = {0};
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_toJSON.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_toJSON.html
index 96c2c42b78..1f3662d9fc 100644
--- a/dom/media/webrtc/tests/mochitests/test_peerConnection_toJSON.html
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_toJSON.html
@@ -28,10 +28,11 @@
                                                    sdpMid: "test",
                                                    sdpMLineIndex: 3 });
     jsonCopy = JSON.parse(JSON.stringify(rtcIceCandidate));
-    for (key in rtcIceCandidate) {
-      if (typeof(rtcIceCandidate[key]) == "function") continue;
-      is(rtcIceCandidate[key], jsonCopy[key], "key " + key + " should match.");
-    }
+    is(jsonCopy.candidate, "dummy");
+    is(jsonCopy.sdpMid, "test");
+    is(jsonCopy.sdpMLineIndex, 3);
+    is(jsonCopy.usernameFragment, rtcIceCandidate.usernameFragment);
+    is(Object.keys(jsonCopy).length, 4, "JSON limited to those four members.");
   });
 </script>
 </pre>
diff --git a/dom/media/webrtc/third_party_build/default_config_env b/dom/media/webrtc/third_party_build/default_config_env
index 7013520a30..be3c5ba7c1 100644
--- a/dom/media/webrtc/third_party_build/default_config_env
+++ b/dom/media/webrtc/third_party_build/default_config_env
@@ -5,41 +5,41 @@
 export MOZ_LIBWEBRTC_SRC=$STATE_DIR/moz-libwebrtc
 
 # The previous fast-forward bug number is used for some error messaging.
-export MOZ_PRIOR_FASTFORWARD_BUG="1871981"
+export MOZ_PRIOR_FASTFORWARD_BUG="1876843"
 
 # Fast-forwarding each Chromium version of libwebrtc should be done
 # under a separate bugzilla bug.  This bug number is used when crafting
 # the commit summary as each upstream commit is vendored into the
 # mercurial repository.  The bug used for the v106 fast-forward was
 # 1800920.
-export MOZ_FASTFORWARD_BUG="1876843"
+export MOZ_FASTFORWARD_BUG="1883116"
 
 # MOZ_NEXT_LIBWEBRTC_MILESTONE and MOZ_NEXT_FIREFOX_REL_TARGET are
 # not used during fast-forward processing, but facilitate generating this
 # default config.  To generate an default config for the next update, run
 # bash dom/media/webrtc/third_party_build/update_default_config_env.sh
-export MOZ_NEXT_LIBWEBRTC_MILESTONE=121
-export MOZ_NEXT_FIREFOX_REL_TARGET=125
+export MOZ_NEXT_LIBWEBRTC_MILESTONE=122
+export MOZ_NEXT_FIREFOX_REL_TARGET=126
 
 # For Chromium release branches, see:
 # https://chromiumdash.appspot.com/branches
 
-# Chromium's v120 release branch was 6099.  This is used to pre-stack
+# Chromium's v121 release branch was 6167.  This is used to pre-stack
 # the previous release branch's commits onto the appropriate base commit
 # (the first common commit between trunk and the release branch).
-export MOZ_PRIOR_UPSTREAM_BRANCH_HEAD_NUM="6099"
+export MOZ_PRIOR_UPSTREAM_BRANCH_HEAD_NUM="6167"
 
-# New target release branch for v121 is branch-heads/6167.  This is used
+# New target release branch for v122 is branch-heads/6261.  This is used
 # to calculate the next upstream commit.
-export MOZ_TARGET_UPSTREAM_BRANCH_HEAD="branch-heads/6167"
+export MOZ_TARGET_UPSTREAM_BRANCH_HEAD="branch-heads/6261"
 
 # For local development 'mozpatches' is fine for a branch name, but when
 # pushing the patch stack to github, it should be named something like
-# 'moz-mods-chr121-for-rel125'.
+# 'moz-mods-chr122-for-rel126'.
 export MOZ_LIBWEBRTC_BRANCH="mozpatches"
 
 # After elm has been merged to mozilla-central, the patch stack in
 # moz-libwebrtc should be pushed to github.  The script
 # push_official_branch.sh uses this branch name when pushing to the
 # public repo.
-export MOZ_LIBWEBRTC_OFFICIAL_BRANCH="moz-mods-chr121-for-rel125"
+export MOZ_LIBWEBRTC_OFFICIAL_BRANCH="moz-mods-chr122-for-rel126"
diff --git a/dom/media/webrtc/third_party_build/elm_rebase.sh b/dom/media/webrtc/third_party_build/elm_rebase.sh
index ba0028b7a4..0dbf93d3ce 100644
--- a/dom/media/webrtc/third_party_build/elm_rebase.sh
+++ b/dom/media/webrtc/third_party_build/elm_rebase.sh
@@ -153,6 +153,15 @@ export MOZ_BOOKMARK=$MOZ_BOOKMARK
 " > $STATE_DIR/rebase_resume_state
 fi # if [ -f $STATE_DIR/rebase_resume_state ]; then ; else
 
+if [ "x$STOP_FOR_REORDER" = "x1" ]; then
+  echo ""
+  echo "Stopping after generating commit list ($COMMIT_LIST_FILE) to"
+  echo "allow tweaking commit ordering.  Re-running $0 will resume the"
+  echo "rebase processing.  To stop processing during the rebase,"
+  echo "insert a line with only 'STOP'."
+  exit
+fi
+
 # grab all commits
 COMMITS=`cat $COMMIT_LIST_FILE | awk '{print $1;}'`
 
@@ -171,6 +180,12 @@ for commit in $COMMITS; do
     ed -s $COMMIT_LIST_FILE <<< $'1d\nw\nq'
   }
 
+  if [ "$FULL_COMMIT_LINE" == "STOP" ]; then
+    echo "Stopping for history editing.  Re-run $0 to resume."
+    remove_commit
+    exit
+  fi
+
   IS_BUILD_COMMIT=`hg log -T '{desc|firstline}' -r $commit \
                    | grep "file updates" | wc -l | tr -d " " || true`
   echo "IS_BUILD_COMMIT: $IS_BUILD_COMMIT"
diff --git a/dom/media/webrtc/third_party_build/fetch_github_repo.py b/dom/media/webrtc/third_party_build/fetch_github_repo.py
index b9d10e0b6c..8caa55d5c5 100644
--- a/dom/media/webrtc/third_party_build/fetch_github_repo.py
+++ b/dom/media/webrtc/third_party_build/fetch_github_repo.py
@@ -87,6 +87,10 @@ def fetch_repo(github_path, clone_protocol, force_fetch, tar_path):
     else:
         print("Upstream remote branch-heads already configured")
 
+    # prevent changing line endings when moving things out of the git repo
+    # (and into hg for instance)
+    run_git("git config --local core.autocrlf false")
+
     # do a sanity fetch in case this was not a freshly cloned copy of the
     # repo, meaning it may not have all the mozilla branches present.
     run_git("git fetch --all", github_path)
diff --git a/dom/media/webrtc/third_party_build/vendor-libwebrtc.py b/dom/media/webrtc/third_party_build/vendor-libwebrtc.py
index d820d8c006..1c44fbd749 100644
--- a/dom/media/webrtc/third_party_build/vendor-libwebrtc.py
+++ b/dom/media/webrtc/third_party_build/vendor-libwebrtc.py
@@ -27,7 +27,6 @@ def get_excluded_files():
         ".clang-format",
         ".git-blame-ignore-revs",
         ".gitignore",
-        ".vpython",
         "CODE_OF_CONDUCT.md",
         "ENG_REVIEW_OWNERS",
         "PRESUBMIT.py",
diff --git a/dom/media/webrtc/transport/test/ice_unittest.cpp b/dom/media/webrtc/transport/test/ice_unittest.cpp
index 50febb3cdd..7df379e1c4 100644
--- a/dom/media/webrtc/transport/test/ice_unittest.cpp
+++ b/dom/media/webrtc/transport/test/ice_unittest.cpp
@@ -58,9 +58,9 @@ using namespace mozilla;
 
 static unsigned int kDefaultTimeout = 7000;
 
-// TODO(nils@mozilla.com): This should get replaced with some non-external
-// solution like discussed in bug 860775.
-const std::string kDefaultStunServerHostname((char*)"stun.l.google.com");
+// TODO: It would be nice to have a test STUN/TURN server that can run with
+// gtest.
+const std::string kDefaultStunServerHostname((char*)"");
 const std::string kBogusStunServerHostname(
     (char*)"stun-server-nonexistent.invalid");
 const uint16_t kDefaultStunServerPort = 19305;
@@ -1628,12 +1628,17 @@ class WebRtcIceConnectTest : public StunTest {
       peer->SetMappingType(mapping_type_);
       peer->SetBlockUdp(block_udp_);
     } else if (setup_stun_servers) {
-      std::vector<NrIceStunServer> stun_servers;
+      if (stun_server_address_.empty()) {
+        InitTestStunServer();
+        peer->UseTestStunServer();
+      } else {
+        std::vector<NrIceStunServer> stun_servers;
 
-      stun_servers.push_back(*NrIceStunServer::Create(
-          stun_server_address_, kDefaultStunServerPort, kNrIceTransportUdp));
+        stun_servers.push_back(*NrIceStunServer::Create(
+            stun_server_address_, kDefaultStunServerPort, kNrIceTransportUdp));
 
-      peer->SetStunServers(stun_servers);
+        peer->SetStunServers(stun_servers);
+      }
     }
   }