mozilla-central: changeset 446960:0ceae9db9ec0be18daa1a279511ad305723185d4

--- a/dom/media/ADTSDecoder.cpp
+++ b/dom/media/ADTSDecoder.cpp
@@ -5,45 +5,41 @@
  * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
 
 #include "ADTSDecoder.h"
 #include "MediaContainerType.h"
 #include "PDMFactory.h"
 
 namespace mozilla {
 
-/* static */ bool
-ADTSDecoder::IsEnabled()
-{
+/* static */ bool ADTSDecoder::IsEnabled() {
   RefPtr<PDMFactory> platform = new PDMFactory();
   return platform->SupportsMimeType(NS_LITERAL_CSTRING("audio/mp4a-latm"),
                                     /* DecoderDoctorDiagnostics* */ nullptr);
 }
 
-/* static */ bool
-ADTSDecoder::IsSupportedType(const MediaContainerType& aContainerType)
-{
+/* static */ bool ADTSDecoder::IsSupportedType(
+    const MediaContainerType& aContainerType) {
   if (aContainerType.Type() == MEDIAMIMETYPE("audio/aac") ||
       aContainerType.Type() == MEDIAMIMETYPE("audio/aacp") ||
       aContainerType.Type() == MEDIAMIMETYPE("audio/x-aac")) {
     return IsEnabled() && (aContainerType.ExtendedType().Codecs().IsEmpty() ||
                            aContainerType.ExtendedType().Codecs() == "aac");
   }
 
   return false;
 }
 
-/* static */ nsTArray<UniquePtr<TrackInfo>>
-ADTSDecoder::GetTracksInfo(const MediaContainerType& aType)
-{
+/* static */ nsTArray<UniquePtr<TrackInfo>> ADTSDecoder::GetTracksInfo(
+    const MediaContainerType& aType) {
   nsTArray<UniquePtr<TrackInfo>> tracks;
   if (!IsSupportedType(aType)) {
     return tracks;
   }
 
   tracks.AppendElement(
-    CreateTrackInfoWithMIMETypeAndContainerTypeExtraParameters(
-      NS_LITERAL_CSTRING("audio/mp4a-latm"), aType));
+      CreateTrackInfoWithMIMETypeAndContainerTypeExtraParameters(
+          NS_LITERAL_CSTRING("audio/mp4a-latm"), aType));
 
   return tracks;
 }
 
-} // namespace mozilla
+}  // namespace mozilla

--- a/dom/media/ADTSDecoder.h
+++ b/dom/media/ADTSDecoder.h
@@ -10,22 +10,21 @@
 #include "mozilla/UniquePtr.h"
 #include "nsTArray.h"
 
 namespace mozilla {
 
 class MediaContainerType;
 class TrackInfo;
 
-class ADTSDecoder
-{
-public:
+class ADTSDecoder {
+ public:
   // Returns true if the ADTS backend is pref'ed on, and we're running on a
   // platform that is likely to have decoders for the format.
   static bool IsEnabled();
   static bool IsSupportedType(const MediaContainerType& aContainerType);
   static nsTArray<UniquePtr<TrackInfo>> GetTracksInfo(
-    const MediaContainerType& aType);
+      const MediaContainerType& aType);
 };
 
-} // namespace mozilla
+}  // namespace mozilla
 
-#endif // !ADTS_DECODER_H_
+#endif  // !ADTS_DECODER_H_

--- a/dom/media/ADTSDemuxer.cpp
+++ b/dom/media/ADTSDemuxer.cpp
@@ -1,36 +1,37 @@
 /* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
 /* vim: set ts=8 sts=2 et sw=2 tw=80: */
 /* This Source Code Form is subject to the terms of the Mozilla Public
-* License, v. 2.0. If a copy of the MPL was not distributed with this
-* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
 
 #include "ADTSDemuxer.h"
 
 #include "TimeUnits.h"
 #include "VideoUtils.h"
 #include "mozilla/UniquePtr.h"
 #include <inttypes.h>
 
 extern mozilla::LazyLogModule gMediaDemuxerLog;
-#define ADTSLOG(msg, ...)                                                      \
+#define ADTSLOG(msg, ...) \
   DDMOZ_LOG(gMediaDemuxerLog, LogLevel::Debug, msg, ##__VA_ARGS__)
-#define ADTSLOGV(msg, ...)                                                     \
+#define ADTSLOGV(msg, ...) \
   DDMOZ_LOG(gMediaDemuxerLog, LogLevel::Verbose, msg, ##__VA_ARGS__)
 
 namespace mozilla {
 namespace adts {
 
 // adts::FrameHeader - Holds the ADTS frame header and its parsing
 // state.
 //
 // ADTS Frame Structure
 //
-// 11111111 1111BCCD EEFFFFGH HHIJKLMM MMMMMMMM MMMOOOOO OOOOOOPP(QQQQQQQQ QQQQQQQQ)
+// 11111111 1111BCCD EEFFFFGH HHIJKLMM MMMMMMMM MMMOOOOO OOOOOOPP(QQQQQQQQ
+// QQQQQQQQ)
 //
 // Header consists of 7 or 9 bytes(without or with CRC).
 // Letter   Length(bits)  Description
 // { sync } 12            syncword 0xFFF, all bits must be 1
 // B        1             MPEG Version: 0 for MPEG-4, 1 for MPEG-2
 // C        2             Layer: always 0
 // D        1             protection absent, Warning, set to 1 if there is no
 //                        CRC and 0 if there is CRC
@@ -41,310 +42,268 @@ namespace adts {
 // M        13            frame length, this value must include 7 or 9 bytes of
 //                        header length: FrameLength =
 //                          (ProtectionAbsent == 1 ? 7 : 9) + size(AACFrame)
 // O        11            Buffer fullness
 // P        2             Number of AAC frames(RDBs) in ADTS frame minus 1, for
 //                        maximum compatibility always use 1 AAC frame per ADTS
 //                        frame
 // Q        16            CRC if protection absent is 0
-class FrameHeader
-{
-public:
+class FrameHeader {
+ public:
   uint32_t mFrameLength;
   uint32_t mSampleRate;
   uint32_t mSamples;
   uint32_t mChannels;
-  uint8_t  mObjectType;
-  uint8_t  mSamplingIndex;
-  uint8_t  mChannelConfig;
-  uint8_t  mNumAACFrames;
-  bool     mHaveCrc;
+  uint8_t mObjectType;
+  uint8_t mSamplingIndex;
+  uint8_t mChannelConfig;
+  uint8_t mNumAACFrames;
+  bool mHaveCrc;
 
   // Returns whether aPtr matches a valid ADTS header sync marker
-  static bool MatchesSync(const uint8_t* aPtr)
-  {
+  static bool MatchesSync(const uint8_t* aPtr) {
     return aPtr[0] == 0xFF && (aPtr[1] & 0xF6) == 0xF0;
   }
 
   FrameHeader() { Reset(); }
 
   // Header size
   size_t HeaderSize() const { return (mHaveCrc) ? 9 : 7; }
 
   bool IsValid() const { return mFrameLength > 0; }
 
   // Resets the state to allow for a new parsing session.
   void Reset() { PodZero(this); }
 
   // Returns whether the byte creates a valid sequence up to this point.
-  bool Parse(const uint8_t* aPtr)
-  {
+  bool Parse(const uint8_t* aPtr) {
     const uint8_t* p = aPtr;
 
     if (!MatchesSync(p)) {
       return false;
     }
 
     // AAC has 1024 samples per frame per channel.
     mSamples = 1024;
 
     mHaveCrc = !(p[1] & 0x01);
     mObjectType = ((p[2] & 0xC0) >> 6) + 1;
     mSamplingIndex = (p[2] & 0x3C) >> 2;
     mChannelConfig = (p[2] & 0x01) << 2 | (p[3] & 0xC0) >> 6;
     mFrameLength =
-      (p[3] & 0x03) << 11 | (p[4] & 0xFF) << 3 | (p[5] & 0xE0) >> 5;
+        (p[3] & 0x03) << 11 | (p[4] & 0xFF) << 3 | (p[5] & 0xE0) >> 5;
     mNumAACFrames = (p[6] & 0x03) + 1;
 
-    static const int32_t SAMPLE_RATES[16] = {
-      96000, 88200, 64000, 48000,
-      44100, 32000, 24000, 22050,
-      16000, 12000, 11025,  8000,
-      7350
-    };
+    static const int32_t SAMPLE_RATES[16] = {96000, 88200, 64000, 48000, 44100,
+                                             32000, 24000, 22050, 16000, 12000,
+                                             11025, 8000,  7350};
     mSampleRate = SAMPLE_RATES[mSamplingIndex];
 
     MOZ_ASSERT(mChannelConfig < 8);
     mChannels = (mChannelConfig == 7) ? 8 : mChannelConfig;
 
     return true;
   }
 };
 
-
 // adts::Frame - Frame meta container used to parse and hold a frame
 // header and side info.
-class Frame
-{
-public:
-  Frame() : mOffset(0), mHeader() { }
+class Frame {
+ public:
+  Frame() : mOffset(0), mHeader() {}
 
   int64_t Offset() const { return mOffset; }
-  size_t Length() const
-  {
+  size_t Length() const {
     // TODO: If fields are zero'd when invalid, this check wouldn't be
     // necessary.
     if (!mHeader.IsValid()) {
       return 0;
     }
 
     return mHeader.mFrameLength;
   }
 
   // Returns the offset to the start of frame's raw data.
   int64_t PayloadOffset() const { return mOffset + mHeader.HeaderSize(); }
 
   // Returns the length of the frame's raw data (excluding the header) in bytes.
-  size_t PayloadLength() const
-  {
+  size_t PayloadLength() const {
     // TODO: If fields are zero'd when invalid, this check wouldn't be
     // necessary.
     if (!mHeader.IsValid()) {
       return 0;
     }
 
     return mHeader.mFrameLength - mHeader.HeaderSize();
   }
 
   // Returns the parsed frame header.
   const FrameHeader& Header() const { return mHeader; }
 
   bool IsValid() const { return mHeader.IsValid(); }
 
   // Resets the frame header and data.
-  void Reset()
-  {
+  void Reset() {
     mHeader.Reset();
     mOffset = 0;
   }
 
   // Returns whether the valid
-  bool Parse(int64_t aOffset, const uint8_t* aStart, const uint8_t* aEnd)
-  {
+  bool Parse(int64_t aOffset, const uint8_t* aStart, const uint8_t* aEnd) {
     MOZ_ASSERT(aStart && aEnd);
 
     bool found = false;
     const uint8_t* ptr = aStart;
     // Require at least 7 bytes of data at the end of the buffer for the minimum
     // ADTS frame header.
     while (ptr < aEnd - 7 && !found) {
       found = mHeader.Parse(ptr);
       ptr++;
     }
 
     mOffset = aOffset + (ptr - aStart) - 1;
 
     return found;
   }
 
-private:
+ private:
   // The offset to the start of the header.
   int64_t mOffset;
 
   // The currently parsed frame header.
   FrameHeader mHeader;
 };
 
-
-class FrameParser
-{
-public:
-
+class FrameParser {
+ public:
   // Returns the currently parsed frame. Reset via Reset or EndFrameSession.
   const Frame& CurrentFrame() const { return mFrame; }
 
-
   // Returns the first parsed frame. Reset via Reset.
   const Frame& FirstFrame() const { return mFirstFrame; }
 
   // Resets the parser. Don't use between frames as first frame data is reset.
-  void Reset()
-  {
+  void Reset() {
     EndFrameSession();
     mFirstFrame.Reset();
   }
 
   // Clear the last parsed frame to allow for next frame parsing, i.e.:
   // - sets PrevFrame to CurrentFrame
   // - resets the CurrentFrame
   // - resets ID3Header if no valid header was parsed yet
-  void EndFrameSession()
-  {
-    mFrame.Reset();
-  }
+  void EndFrameSession() { mFrame.Reset(); }
 
   // Parses contents of given ByteReader for a valid frame header and returns
   // true if one was found. After returning, the variable passed to
   // 'aBytesToSkip' holds the amount of bytes to be skipped (if any) in order to
   // jump across a large ID3v2 tag spanning multiple buffers.
-  bool Parse(int64_t aOffset, const uint8_t* aStart, const uint8_t* aEnd)
-  {
+  bool Parse(int64_t aOffset, const uint8_t* aStart, const uint8_t* aEnd) {
     const bool found = mFrame.Parse(aOffset, aStart, aEnd);
 
     if (mFrame.Length() && !mFirstFrame.Length()) {
       mFirstFrame = mFrame;
     }
 
     return found;
   }
 
-private:
+ private:
   // We keep the first parsed frame around for static info access, the
   // previously parsed frame for debugging and the currently parsed frame.
   Frame mFirstFrame;
   Frame mFrame;
 };
 
 // Initialize the AAC AudioSpecificConfig.
 // Only handles two-byte version for AAC-LC.
-static void
-InitAudioSpecificConfig(const Frame& frame,
-                        MediaByteBuffer* aBuffer)
-{
+static void InitAudioSpecificConfig(const Frame& frame,
+                                    MediaByteBuffer* aBuffer) {
   const FrameHeader& header = frame.Header();
   MOZ_ASSERT(header.IsValid());
 
   int audioObjectType = header.mObjectType;
   int samplingFrequencyIndex = header.mSamplingIndex;
   int channelConfig = header.mChannelConfig;
 
   uint8_t asc[2];
   asc[0] = (audioObjectType & 0x1F) << 3 | (samplingFrequencyIndex & 0x0E) >> 1;
   asc[1] = (samplingFrequencyIndex & 0x01) << 7 | (channelConfig & 0x0F) << 3;
 
   aBuffer->AppendElements(asc, 2);
 }
 
-} // namespace adts
+}  // namespace adts
 
 using media::TimeUnit;
 
 // ADTSDemuxer
 
-ADTSDemuxer::ADTSDemuxer(MediaResource* aSource)
-  : mSource(aSource)
-{
+ADTSDemuxer::ADTSDemuxer(MediaResource* aSource) : mSource(aSource) {
   DDLINKCHILD("source", aSource);
 }
 
-bool
-ADTSDemuxer::InitInternal()
-{
+bool ADTSDemuxer::InitInternal() {
   if (!mTrackDemuxer) {
     mTrackDemuxer = new ADTSTrackDemuxer(mSource);
     DDLINKCHILD("track demuxer", mTrackDemuxer.get());
   }
   return mTrackDemuxer->Init();
 }
 
-RefPtr<ADTSDemuxer::InitPromise>
-ADTSDemuxer::Init()
-{
+RefPtr<ADTSDemuxer::InitPromise> ADTSDemuxer::Init() {
   if (!InitInternal()) {
     ADTSLOG("Init() failure: waiting for data");
 
-    return InitPromise::CreateAndReject(
-      NS_ERROR_DOM_MEDIA_METADATA_ERR, __func__);
+    return InitPromise::CreateAndReject(NS_ERROR_DOM_MEDIA_METADATA_ERR,
+                                        __func__);
   }
 
   ADTSLOG("Init() successful");
   return InitPromise::CreateAndResolve(NS_OK, __func__);
 }
 
-uint32_t
-ADTSDemuxer::GetNumberTracks(TrackInfo::TrackType aType) const
-{
+uint32_t ADTSDemuxer::GetNumberTracks(TrackInfo::TrackType aType) const {
   return (aType == TrackInfo::kAudioTrack) ? 1 : 0;
 }
 
-already_AddRefed<MediaTrackDemuxer>
-ADTSDemuxer::GetTrackDemuxer(TrackInfo::TrackType aType, uint32_t aTrackNumber)
-{
+already_AddRefed<MediaTrackDemuxer> ADTSDemuxer::GetTrackDemuxer(
+    TrackInfo::TrackType aType, uint32_t aTrackNumber) {
   if (!mTrackDemuxer) {
     return nullptr;
   }
 
   return RefPtr<ADTSTrackDemuxer>(mTrackDemuxer).forget();
 }
 
-bool
-ADTSDemuxer::IsSeekable() const
-{
+bool ADTSDemuxer::IsSeekable() const {
   int64_t length = mSource->GetLength();
-  if (length > -1)
-    return true;
+  if (length > -1) return true;
   return false;
 }
 
-
 // ADTSTrackDemuxer
 ADTSTrackDemuxer::ADTSTrackDemuxer(MediaResource* aSource)
-  : mSource(aSource)
-  , mParser(new adts::FrameParser())
-  , mOffset(0)
-  , mNumParsedFrames(0)
-  , mFrameIndex(0)
-  , mTotalFrameLen(0)
-  , mSamplesPerFrame(0)
-  , mSamplesPerSecond(0)
-  , mChannels(0)
-{
+    : mSource(aSource),
+      mParser(new adts::FrameParser()),
+      mOffset(0),
+      mNumParsedFrames(0),
+      mFrameIndex(0),
+      mTotalFrameLen(0),
+      mSamplesPerFrame(0),
+      mSamplesPerSecond(0),
+      mChannels(0) {
   DDLINKCHILD("source", aSource);
   Reset();
 }
 
-ADTSTrackDemuxer::~ADTSTrackDemuxer()
-{
-  delete mParser;
-}
+ADTSTrackDemuxer::~ADTSTrackDemuxer() { delete mParser; }
 
-bool
-ADTSTrackDemuxer::Init()
-{
+bool ADTSTrackDemuxer::Init() {
   FastSeek(TimeUnit::Zero());
   // Read the first frame to fetch sample rate and other meta data.
   RefPtr<MediaRawData> frame(GetNextFrame(FindNextFrame(true)));
 
   ADTSLOG("Init StreamLength()=%" PRId64 " first-frame-found=%d",
           StreamLength(), !!frame);
 
   if (!frame) {
@@ -365,228 +324,202 @@ ADTSTrackDemuxer::Init()
 
   // AAC Specific information
   mInfo->mMimeType = "audio/mp4a-latm";
 
   // Configure AAC codec-specific values.
   // For AAC, mProfile and mExtendedProfile contain the audioObjectType from
   // Table 1.3 -- Audio Profile definition, ISO/IEC 14496-3. Eg. 2 == AAC LC
   mInfo->mProfile = mInfo->mExtendedProfile =
-    mParser->FirstFrame().Header().mObjectType;
+      mParser->FirstFrame().Header().mObjectType;
   InitAudioSpecificConfig(mParser->FirstFrame(), mInfo->mCodecSpecificConfig);
 
   ADTSLOG("Init mInfo={mRate=%u mChannels=%u mBitDepth=%u mDuration=%" PRId64
           "}",
           mInfo->mRate, mInfo->mChannels, mInfo->mBitDepth,
           mInfo->mDuration.ToMicroseconds());
 
   return mSamplesPerSecond && mChannels;
 }
 
-UniquePtr<TrackInfo>
-ADTSTrackDemuxer::GetInfo() const
-{
+UniquePtr<TrackInfo> ADTSTrackDemuxer::GetInfo() const {
   return mInfo->Clone();
 }
 
-RefPtr<ADTSTrackDemuxer::SeekPromise>
-ADTSTrackDemuxer::Seek(const TimeUnit& aTime)
-{
+RefPtr<ADTSTrackDemuxer::SeekPromise> ADTSTrackDemuxer::Seek(
+    const TimeUnit& aTime) {
   // Efficiently seek to the position.
   FastSeek(aTime);
   // Correct seek position by scanning the next frames.
   const TimeUnit seekTime = ScanUntil(aTime);
 
   return SeekPromise::CreateAndResolve(seekTime, __func__);
 }
 
-TimeUnit
-ADTSTrackDemuxer::FastSeek(const TimeUnit& aTime)
-{
+TimeUnit ADTSTrackDemuxer::FastSeek(const TimeUnit& aTime) {
   ADTSLOG("FastSeek(%" PRId64 ") avgFrameLen=%f mNumParsedFrames=%" PRIu64
-         " mFrameIndex=%" PRId64 " mOffset=%" PRIu64,
-         aTime.ToMicroseconds(), AverageFrameLength(), mNumParsedFrames,
-         mFrameIndex, mOffset);
+          " mFrameIndex=%" PRId64 " mOffset=%" PRIu64,
+          aTime.ToMicroseconds(), AverageFrameLength(), mNumParsedFrames,
+          mFrameIndex, mOffset);
 
   const int64_t firstFrameOffset = mParser->FirstFrame().Offset();
   if (!aTime.ToMicroseconds()) {
     // Quick seek to the beginning of the stream.
     mOffset = firstFrameOffset;
   } else if (AverageFrameLength() > 0) {
-    mOffset = firstFrameOffset + FrameIndexFromTime(aTime) *
-      AverageFrameLength();
+    mOffset =
+        firstFrameOffset + FrameIndexFromTime(aTime) * AverageFrameLength();
   }
 
   if (mOffset > firstFrameOffset && StreamLength() > 0) {
     mOffset = std::min(StreamLength() - 1, mOffset);
   }
 
   mFrameIndex = FrameIndexFromOffset(mOffset);
   mParser->EndFrameSession();
 
   ADTSLOG("FastSeek End avgFrameLen=%f mNumParsedFrames=%" PRIu64
-          " mFrameIndex=%" PRId64 " mFirstFrameOffset=%" PRIu64 " mOffset=%" PRIu64
-          " SL=%" PRIu64 "",
-          AverageFrameLength(), mNumParsedFrames, mFrameIndex,
-          firstFrameOffset, mOffset, StreamLength());
+          " mFrameIndex=%" PRId64 " mFirstFrameOffset=%" PRIu64
+          " mOffset=%" PRIu64 " SL=%" PRIu64 "",
+          AverageFrameLength(), mNumParsedFrames, mFrameIndex, firstFrameOffset,
+          mOffset, StreamLength());
 
   return Duration(mFrameIndex);
 }
 
-TimeUnit
-ADTSTrackDemuxer::ScanUntil(const TimeUnit& aTime)
-{
+TimeUnit ADTSTrackDemuxer::ScanUntil(const TimeUnit& aTime) {
   ADTSLOG("ScanUntil(%" PRId64 ") avgFrameLen=%f mNumParsedFrames=%" PRIu64
           " mFrameIndex=%" PRId64 " mOffset=%" PRIu64,
           aTime.ToMicroseconds(), AverageFrameLength(), mNumParsedFrames,
           mFrameIndex, mOffset);
 
   if (!aTime.ToMicroseconds()) {
     return FastSeek(aTime);
   }
 
   if (Duration(mFrameIndex) > aTime) {
     FastSeek(aTime);
   }
 
   while (SkipNextFrame(FindNextFrame()) && Duration(mFrameIndex + 1) < aTime) {
     ADTSLOGV("ScanUntil* avgFrameLen=%f mNumParsedFrames=%" PRIu64
              " mFrameIndex=%" PRId64 " mOffset=%" PRIu64 " Duration=%" PRId64,
-             AverageFrameLength(), mNumParsedFrames, mFrameIndex,
-             mOffset, Duration(mFrameIndex + 1).ToMicroseconds());
+             AverageFrameLength(), mNumParsedFrames, mFrameIndex, mOffset,
+             Duration(mFrameIndex + 1).ToMicroseconds());
   }
 
   ADTSLOG("ScanUntil End avgFrameLen=%f mNumParsedFrames=%" PRIu64
           " mFrameIndex=%" PRId64 " mOffset=%" PRIu64,
           AverageFrameLength(), mNumParsedFrames, mFrameIndex, mOffset);
 
   return Duration(mFrameIndex);
 }
 
-RefPtr<ADTSTrackDemuxer::SamplesPromise>
-ADTSTrackDemuxer::GetSamples(int32_t aNumSamples)
-{
+RefPtr<ADTSTrackDemuxer::SamplesPromise> ADTSTrackDemuxer::GetSamples(
+    int32_t aNumSamples) {
   ADTSLOGV("GetSamples(%d) Begin mOffset=%" PRIu64 " mNumParsedFrames=%" PRIu64
            " mFrameIndex=%" PRId64 " mTotalFrameLen=%" PRIu64
            " mSamplesPerFrame=%d "
            "mSamplesPerSecond=%d mChannels=%d",
            aNumSamples, mOffset, mNumParsedFrames, mFrameIndex, mTotalFrameLen,
            mSamplesPerFrame, mSamplesPerSecond, mChannels);
 
   MOZ_ASSERT(aNumSamples);
 
   RefPtr<SamplesHolder> frames = new SamplesHolder();
 
   while (aNumSamples--) {
     RefPtr<MediaRawData> frame(GetNextFrame(FindNextFrame()));
-    if (!frame)
-      break;
+    if (!frame) break;
 
     frames->mSamples.AppendElement(frame);
   }
 
-  ADTSLOGV("GetSamples() End mSamples.Size()=%zu aNumSamples=%d mOffset=%" PRIu64
-           " mNumParsedFrames=%" PRIu64 " mFrameIndex=%" PRId64
-           " mTotalFrameLen=%" PRIu64
-           " mSamplesPerFrame=%d mSamplesPerSecond=%d "
-           "mChannels=%d",
-           frames->mSamples.Length(), aNumSamples, mOffset, mNumParsedFrames,
-           mFrameIndex, mTotalFrameLen, mSamplesPerFrame, mSamplesPerSecond,
-           mChannels);
+  ADTSLOGV(
+      "GetSamples() End mSamples.Size()=%zu aNumSamples=%d mOffset=%" PRIu64
+      " mNumParsedFrames=%" PRIu64 " mFrameIndex=%" PRId64
+      " mTotalFrameLen=%" PRIu64
+      " mSamplesPerFrame=%d mSamplesPerSecond=%d "
+      "mChannels=%d",
+      frames->mSamples.Length(), aNumSamples, mOffset, mNumParsedFrames,
+      mFrameIndex, mTotalFrameLen, mSamplesPerFrame, mSamplesPerSecond,
+      mChannels);
 
   if (frames->mSamples.IsEmpty()) {
-    return SamplesPromise::CreateAndReject(
-      NS_ERROR_DOM_MEDIA_END_OF_STREAM, __func__);
+    return SamplesPromise::CreateAndReject(NS_ERROR_DOM_MEDIA_END_OF_STREAM,
+                                           __func__);
   }
 
   return SamplesPromise::CreateAndResolve(frames, __func__);
 }
 
-void
-ADTSTrackDemuxer::Reset()
-{
+void ADTSTrackDemuxer::Reset() {
   ADTSLOG("Reset()");
   MOZ_ASSERT(mParser);
   if (mParser) {
     mParser->Reset();
   }
   FastSeek(TimeUnit::Zero());
 }
 
 RefPtr<ADTSTrackDemuxer::SkipAccessPointPromise>
-ADTSTrackDemuxer::SkipToNextRandomAccessPoint(const TimeUnit& aTimeThreshold)
-{
+ADTSTrackDemuxer::SkipToNextRandomAccessPoint(const TimeUnit& aTimeThreshold) {
   // Will not be called for audio-only resources.
   return SkipAccessPointPromise::CreateAndReject(
-    SkipFailureHolder(NS_ERROR_DOM_MEDIA_DEMUXER_ERR, 0), __func__);
+      SkipFailureHolder(NS_ERROR_DOM_MEDIA_DEMUXER_ERR, 0), __func__);
 }
 
-int64_t
-ADTSTrackDemuxer::GetResourceOffset() const
-{
-  return mOffset;
-}
+int64_t ADTSTrackDemuxer::GetResourceOffset() const { return mOffset; }
 
-media::TimeIntervals
-ADTSTrackDemuxer::GetBuffered()
-{
+media::TimeIntervals ADTSTrackDemuxer::GetBuffered() {
   auto duration = Duration();
 
   if (!duration.IsPositive()) {
     return media::TimeIntervals();
   }
 
   AutoPinned<MediaResource> stream(mSource.GetResource());
   return GetEstimatedBufferedTimeRanges(stream, duration.ToMicroseconds());
 }
 
-int64_t
-ADTSTrackDemuxer::StreamLength() const
-{
-  return mSource.GetLength();
-}
+int64_t ADTSTrackDemuxer::StreamLength() const { return mSource.GetLength(); }
 
-TimeUnit
-ADTSTrackDemuxer::Duration() const
-{
+TimeUnit ADTSTrackDemuxer::Duration() const {
   if (!mNumParsedFrames) {
     return TimeUnit::FromMicroseconds(-1);
   }
 
   const int64_t streamLen = StreamLength();
   if (streamLen < 0) {
     // Unknown length, we can't estimate duration.
     return TimeUnit::FromMicroseconds(-1);
   }
   const int64_t firstFrameOffset = mParser->FirstFrame().Offset();
   int64_t numFrames = (streamLen - firstFrameOffset) / AverageFrameLength();
   return Duration(numFrames);
 }
 
-TimeUnit
-ADTSTrackDemuxer::Duration(int64_t aNumFrames) const
-{
+TimeUnit ADTSTrackDemuxer::Duration(int64_t aNumFrames) const {
   if (!mSamplesPerSecond) {
     return TimeUnit::FromMicroseconds(-1);
   }
 
   return FramesToTimeUnit(aNumFrames * mSamplesPerFrame, mSamplesPerSecond);
 }
 
-const adts::Frame&
-ADTSTrackDemuxer::FindNextFrame(bool findFirstFrame /*= false*/)
-{
+const adts::Frame& ADTSTrackDemuxer::FindNextFrame(
+    bool findFirstFrame /*= false*/) {
   static const int BUFFER_SIZE = 4096;
   static const int MAX_SKIPPED_BYTES = 10 * BUFFER_SIZE;
 
   ADTSLOGV("FindNext() Begin mOffset=%" PRIu64 " mNumParsedFrames=%" PRIu64
-          " mFrameIndex=%" PRId64 " mTotalFrameLen=%" PRIu64
-          " mSamplesPerFrame=%d mSamplesPerSecond=%d mChannels=%d",
-          mOffset, mNumParsedFrames, mFrameIndex, mTotalFrameLen,
-          mSamplesPerFrame, mSamplesPerSecond, mChannels);
+           " mFrameIndex=%" PRId64 " mTotalFrameLen=%" PRIu64
+           " mSamplesPerFrame=%d mSamplesPerSecond=%d mChannels=%d",
+           mOffset, mNumParsedFrames, mFrameIndex, mTotalFrameLen,
+           mSamplesPerFrame, mSamplesPerSecond, mChannels);
 
   uint8_t buffer[BUFFER_SIZE];
   int32_t read = 0;
 
   bool foundFrame = false;
   int64_t frameHeaderOffset = mOffset;
 
   // Prepare the parser for the next frame parsing session.
@@ -607,17 +540,17 @@ ADTSTrackDemuxer::FindNextFrame(bool fin
     const adts::Frame& currentFrame = mParser->CurrentFrame();
     foundFrame = mParser->Parse(frameHeaderOffset, buffer, buffer + read);
     if (findFirstFrame && foundFrame) {
       // Check for sync marker after the found frame, since it's
       // possible to find sync marker in AAC data. If sync marker
       // exists after the current frame then we've found a frame
       // header.
       int64_t nextFrameHeaderOffset =
-        currentFrame.Offset() + currentFrame.Length();
+          currentFrame.Offset() + currentFrame.Length();
       int32_t read = Read(buffer, nextFrameHeaderOffset, 2);
       if (read != 2 || !adts::FrameHeader::MatchesSync(buffer)) {
         frameHeaderOffset = currentFrame.Offset() + 1;
         mParser->Reset();
         foundFrame = false;
         continue;
       }
     }
@@ -633,59 +566,59 @@ ADTSTrackDemuxer::FindNextFrame(bool fin
     if (frameHeaderOffset + advance <= frameHeaderOffset) {
       break;
     }
 
     frameHeaderOffset += advance;
   }
 
   if (!foundFrame || !mParser->CurrentFrame().Length()) {
-    ADTSLOG("FindNext() Exit foundFrame=%d mParser->CurrentFrame().Length()=%zu ",
-           foundFrame, mParser->CurrentFrame().Length());
+    ADTSLOG(
+        "FindNext() Exit foundFrame=%d mParser->CurrentFrame().Length()=%zu ",
+        foundFrame, mParser->CurrentFrame().Length());
     mParser->Reset();
     return mParser->CurrentFrame();
   }
 
   ADTSLOGV("FindNext() End mOffset=%" PRIu64 " mNumParsedFrames=%" PRIu64
-          " mFrameIndex=%" PRId64 " frameHeaderOffset=%" PRId64
-          " mTotalFrameLen=%" PRIu64 " mSamplesPerFrame=%d mSamplesPerSecond=%d"
-          " mChannels=%d",
-          mOffset, mNumParsedFrames, mFrameIndex, frameHeaderOffset,
-          mTotalFrameLen, mSamplesPerFrame, mSamplesPerSecond, mChannels);
+           " mFrameIndex=%" PRId64 " frameHeaderOffset=%" PRId64
+           " mTotalFrameLen=%" PRIu64
+           " mSamplesPerFrame=%d mSamplesPerSecond=%d"
+           " mChannels=%d",
+           mOffset, mNumParsedFrames, mFrameIndex, frameHeaderOffset,
+           mTotalFrameLen, mSamplesPerFrame, mSamplesPerSecond, mChannels);
 
   return mParser->CurrentFrame();
 }
 
-bool
-ADTSTrackDemuxer::SkipNextFrame(const adts::Frame& aFrame)
-{
+bool ADTSTrackDemuxer::SkipNextFrame(const adts::Frame& aFrame) {
   if (!mNumParsedFrames || !aFrame.Length()) {
     RefPtr<MediaRawData> frame(GetNextFrame(aFrame));
     return frame;
   }
 
   UpdateState(aFrame);
 
   ADTSLOGV("SkipNext() End mOffset=%" PRIu64 " mNumParsedFrames=%" PRIu64
-          " mFrameIndex=%" PRId64 " mTotalFrameLen=%" PRIu64
-          " mSamplesPerFrame=%d mSamplesPerSecond=%d mChannels=%d",
-          mOffset, mNumParsedFrames, mFrameIndex, mTotalFrameLen,
-          mSamplesPerFrame, mSamplesPerSecond, mChannels);
+           " mFrameIndex=%" PRId64 " mTotalFrameLen=%" PRIu64
+           " mSamplesPerFrame=%d mSamplesPerSecond=%d mChannels=%d",
+           mOffset, mNumParsedFrames, mFrameIndex, mTotalFrameLen,
+           mSamplesPerFrame, mSamplesPerSecond, mChannels);
 
   return true;
 }
 
-already_AddRefed<MediaRawData>
-ADTSTrackDemuxer::GetNextFrame(const adts::Frame& aFrame)
-{
-  ADTSLOG("GetNext() Begin({mOffset=%" PRId64 " HeaderSize()=%zu"
+already_AddRefed<MediaRawData> ADTSTrackDemuxer::GetNextFrame(
+    const adts::Frame& aFrame) {
+  ADTSLOG("GetNext() Begin({mOffset=%" PRId64
+          " HeaderSize()=%zu"
           " Length()=%zu})",
-         aFrame.Offset(), aFrame.Header().HeaderSize(), aFrame.PayloadLength());
-  if (!aFrame.IsValid())
-    return nullptr;
+          aFrame.Offset(), aFrame.Header().HeaderSize(),
+          aFrame.PayloadLength());
+  if (!aFrame.IsValid()) return nullptr;
 
   const int64_t offset = aFrame.PayloadOffset();
   const uint32_t length = aFrame.PayloadLength();
 
   RefPtr<MediaRawData> frame = new MediaRawData();
   frame->mOffset = offset;
 
   UniquePtr<MediaRawDataWriter> frameWriter(frame->CreateWriter());
@@ -714,46 +647,41 @@ ADTSTrackDemuxer::GetNextFrame(const adt
            " mFrameIndex=%" PRId64 " mTotalFrameLen=%" PRIu64
            " mSamplesPerFrame=%d mSamplesPerSecond=%d mChannels=%d",
            mOffset, mNumParsedFrames, mFrameIndex, mTotalFrameLen,
            mSamplesPerFrame, mSamplesPerSecond, mChannels);
 
   return frame.forget();
 }
 
-int64_t
-ADTSTrackDemuxer::FrameIndexFromOffset(int64_t aOffset) const
-{
+int64_t ADTSTrackDemuxer::FrameIndexFromOffset(int64_t aOffset) const {
   int64_t frameIndex = 0;
 
   if (AverageFrameLength() > 0) {
     frameIndex =
-      (aOffset - mParser->FirstFrame().Offset()) / AverageFrameLength();
+        (aOffset - mParser->FirstFrame().Offset()) / AverageFrameLength();
   }
 
-  ADTSLOGV("FrameIndexFromOffset(%" PRId64 ") -> %" PRId64, aOffset, frameIndex);
+  ADTSLOGV("FrameIndexFromOffset(%" PRId64 ") -> %" PRId64, aOffset,
+           frameIndex);
   return std::max<int64_t>(0, frameIndex);
 }
 
-int64_t
-ADTSTrackDemuxer::FrameIndexFromTime(const TimeUnit& aTime) const
-{
+int64_t ADTSTrackDemuxer::FrameIndexFromTime(const TimeUnit& aTime) const {
   int64_t frameIndex = 0;
   if (mSamplesPerSecond > 0 && mSamplesPerFrame > 0) {
     frameIndex = aTime.ToSeconds() * mSamplesPerSecond / mSamplesPerFrame - 1;
   }
 
-  ADTSLOGV("FrameIndexFromOffset(%fs) -> %" PRId64,
-           aTime.ToSeconds(), frameIndex);
+  ADTSLOGV("FrameIndexFromOffset(%fs) -> %" PRId64, aTime.ToSeconds(),
+           frameIndex);
   return std::max<int64_t>(0, frameIndex);
 }
 
-void
-ADTSTrackDemuxer::UpdateState(const adts::Frame& aFrame)
-{
+void ADTSTrackDemuxer::UpdateState(const adts::Frame& aFrame) {
   int32_t frameLength = aFrame.Length();
   // Prevent overflow.
   if (mTotalFrameLen + frameLength < mTotalFrameLen) {
     // These variables have a linear dependency and are only used to derive the
     // average frame length.
     mTotalFrameLen /= 2;
     mNumParsedFrames /= 2;
   }
@@ -769,49 +697,45 @@ ADTSTrackDemuxer::UpdateState(const adts
     mChannels = header.mChannels;
   }
 
   ++mNumParsedFrames;
   ++mFrameIndex;
   MOZ_ASSERT(mFrameIndex > 0);
 }
 
-int32_t
-ADTSTrackDemuxer::Read(uint8_t* aBuffer, int64_t aOffset, int32_t aSize)
-{
-  ADTSLOGV("ADTSTrackDemuxer::Read(%p %" PRId64 " %d)",
-           aBuffer, aOffset, aSize);
+int32_t ADTSTrackDemuxer::Read(uint8_t* aBuffer, int64_t aOffset,
+                               int32_t aSize) {
+  ADTSLOGV("ADTSTrackDemuxer::Read(%p %" PRId64 " %d)", aBuffer, aOffset,
+           aSize);
 
   const int64_t streamLen = StreamLength();
   if (mInfo && streamLen > 0) {
     // Prevent blocking reads after successful initialization.
     aSize = std::min<int64_t>(aSize, streamLen - aOffset);
   }
 
   uint32_t read = 0;
   ADTSLOGV("ADTSTrackDemuxer::Read        -> ReadAt(%d)", aSize);
   const nsresult rv = mSource.ReadAt(aOffset, reinterpret_cast<char*>(aBuffer),
                                      static_cast<uint32_t>(aSize), &read);
   NS_ENSURE_SUCCESS(rv, 0);
   return static_cast<int32_t>(read);
 }
 
-double
-ADTSTrackDemuxer::AverageFrameLength() const
-{
+double ADTSTrackDemuxer::AverageFrameLength() const {
   if (mNumParsedFrames) {
     return static_cast<double>(mTotalFrameLen) / mNumParsedFrames;
   }
 
   return 0.0;
 }
 
-/* static */ bool
-ADTSDemuxer::ADTSSniffer(const uint8_t* aData, const uint32_t aLength)
-{
+/* static */ bool ADTSDemuxer::ADTSSniffer(const uint8_t* aData,
+                                           const uint32_t aLength) {
   if (aLength < 7) {
     return false;
   }
   if (!adts::FrameHeader::MatchesSync(aData)) {
     return false;
   }
   auto parser = MakeUnique<adts::FrameParser>();
 
@@ -824,9 +748,9 @@ ADTSDemuxer::ADTSSniffer(const uint8_t* 
   // exists after the current frame then we've found a frame
   // header.
   int64_t nextFrameHeaderOffset = currentFrame.Offset() + currentFrame.Length();
   return int64_t(aLength) > nextFrameHeaderOffset &&
          aLength - nextFrameHeaderOffset >= 2 &&
          adts::FrameHeader::MatchesSync(aData + nextFrameHeaderOffset);
 }
 
-} // namespace mozilla
+}  // namespace mozilla

--- a/dom/media/ADTSDemuxer.h
+++ b/dom/media/ADTSDemuxer.h
@@ -1,63 +1,59 @@
 /* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
 /* vim: set ts=8 sts=2 et sw=2 tw=80: */
 /* This Source Code Form is subject to the terms of the Mozilla Public
-* License, v. 2.0. If a copy of the MPL was not distributed with this
-* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
 
 #ifndef ADTS_DEMUXER_H_
 #define ADTS_DEMUXER_H_
 
 #include "mozilla/Attributes.h"
 #include "mozilla/Maybe.h"
 #include "MediaDataDemuxer.h"
 #include "MediaResource.h"
 
 namespace mozilla {
 
 namespace adts {
 class Frame;
 class FrameParser;
-}
+}  // namespace adts
 
 class ADTSTrackDemuxer;
 
 DDLoggedTypeDeclNameAndBase(ADTSDemuxer, MediaDataDemuxer);
 
-class ADTSDemuxer
-  : public MediaDataDemuxer
-  , public DecoderDoctorLifeLogger<ADTSDemuxer>
-{
-public:
+class ADTSDemuxer : public MediaDataDemuxer,
+                    public DecoderDoctorLifeLogger<ADTSDemuxer> {
+ public:
   // MediaDataDemuxer interface.
   explicit ADTSDemuxer(MediaResource* aSource);
   RefPtr<InitPromise> Init() override;
   uint32_t GetNumberTracks(TrackInfo::TrackType aType) const override;
-  already_AddRefed<MediaTrackDemuxer>
-  GetTrackDemuxer(TrackInfo::TrackType aType, uint32_t aTrackNumber) override;
+  already_AddRefed<MediaTrackDemuxer> GetTrackDemuxer(
+      TrackInfo::TrackType aType, uint32_t aTrackNumber) override;
   bool IsSeekable() const override;
 
   // Return true if a valid ADTS frame header could be found.
   static bool ADTSSniffer(const uint8_t* aData, const uint32_t aLength);
 
-private:
+ private:
   bool InitInternal();
 
   RefPtr<MediaResource> mSource;
   RefPtr<ADTSTrackDemuxer> mTrackDemuxer;
 };
 
 DDLoggedTypeNameAndBase(ADTSTrackDemuxer, MediaTrackDemuxer);
 
-class ADTSTrackDemuxer
-  : public MediaTrackDemuxer
-  , public DecoderDoctorLifeLogger<ADTSTrackDemuxer>
-{
-public:
+class ADTSTrackDemuxer : public MediaTrackDemuxer,
+                         public DecoderDoctorLifeLogger<ADTSTrackDemuxer> {
+ public:
   explicit ADTSTrackDemuxer(MediaResource* aSource);
 
   // Initializes the track demuxer by reading the first frame for meta data.
   // Returns initialization success state.
   bool Init();
 
   // Returns the total stream length if known, -1 otherwise.
   int64_t StreamLength() const;
@@ -70,28 +66,29 @@ public:
   media::TimeUnit Duration(int64_t aNumFrames) const;
 
   // MediaTrackDemuxer interface.
   UniquePtr<TrackInfo> GetInfo() const override;
   RefPtr<SeekPromise> Seek(const media::TimeUnit& aTime) override;
   RefPtr<SamplesPromise> GetSamples(int32_t aNumSamples = 1) override;
   void Reset() override;
   RefPtr<SkipAccessPointPromise> SkipToNextRandomAccessPoint(
-    const media::TimeUnit& aTimeThreshold) override;
+      const media::TimeUnit& aTimeThreshold) override;
   int64_t GetResourceOffset() const override;
   media::TimeIntervals GetBuffered() override;
 
-private:
+ private:
   // Destructor.
   ~ADTSTrackDemuxer();
 
   // Fast approximate seeking to given time.
   media::TimeUnit FastSeek(const media::TimeUnit& aTime);
 
-  // Seeks by scanning the stream up to the given time for more accurate results.
+  // Seeks by scanning the stream up to the given time for more accurate
+  // results.
   media::TimeUnit ScanUntil(const media::TimeUnit& aTime);
 
   // Finds the next valid frame and returns its byte range.
   const adts::Frame& FindNextFrame(bool findFirstFrame = false);
 
   // Skips the next frame given the provided byte range.
   bool SkipNextFrame(const adts::Frame& aFrame);
 
@@ -130,21 +127,22 @@ private:
   int64_t mFrameIndex;
 
   // Sum of parsed frames' lengths in bytes.
   uint64_t mTotalFrameLen;
 
   // Samples per frame metric derived from frame headers or 0 if none available.
   uint32_t mSamplesPerFrame;
 
-  // Samples per second metric derived from frame headers or 0 if none available.
+  // Samples per second metric derived from frame headers or 0 if none
+  // available.
   uint32_t mSamplesPerSecond;
 
   // Channel count derived from frame headers or 0 if none available.
   uint32_t mChannels;
 
   // Audio track config info.
   UniquePtr<AudioInfo> mInfo;
 };
 
-} // mozilla
+}  // namespace mozilla
 
-#endif // !ADTS_DEMUXER_H_
+#endif  // !ADTS_DEMUXER_H_

--- a/dom/media/AsyncLogger.h
+++ b/dom/media/AsyncLogger.h
@@ -13,242 +13,223 @@
 #include "mozilla/Attributes.h"
 #include "mozilla/MathAlgorithms.h"
 #include "mozilla/Sprintf.h"
 
 namespace mozilla {
 
 namespace detail {
 
- // This class implements a lock-free multiple producer single consumer queue of
- // fixed size log messages, with the following characteristics:
- // - Unbounded (uses a intrinsic linked list)
- // - Allocates on Push. Push can be called on any thread.
- // - Deallocates on Pop. Pop MUST always be called on the same thread for the
- // life-time of the queue.
- //
- // In our scenario, the producer threads are real-time, they can't block. The
- // consummer thread runs every now and then and empties the queue to a log
- // file, on disk.
- //
- // Having fixed size messages and jemalloc is probably not the fastest, but
- // allows having a simpler design, we count on the fact that jemalloc will get
- // the memory from a thread-local source most of the time.
-template<size_t MESSAGE_LENGTH>
-class MPSCQueue
-{
-public:
-    struct Message {
-        Message()
-        {
-           mNext.store(nullptr, std::memory_order_relaxed);
-        }
-        Message(const Message& aMessage) = delete;
-        void operator=(const Message& aMessage) = delete;
+// This class implements a lock-free multiple producer single consumer queue of
+// fixed size log messages, with the following characteristics:
+// - Unbounded (uses a intrinsic linked list)
+// - Allocates on Push. Push can be called on any thread.
+// - Deallocates on Pop. Pop MUST always be called on the same thread for the
+// life-time of the queue.
+//
+// In our scenario, the producer threads are real-time, they can't block. The
+// consummer thread runs every now and then and empties the queue to a log
+// file, on disk.
+//
+// Having fixed size messages and jemalloc is probably not the fastest, but
+// allows having a simpler design, we count on the fact that jemalloc will get
+// the memory from a thread-local source most of the time.
+template <size_t MESSAGE_LENGTH>
+class MPSCQueue {
+ public:
+  struct Message {
+    Message() { mNext.store(nullptr, std::memory_order_relaxed); }
+    Message(const Message& aMessage) = delete;
+    void operator=(const Message& aMessage) = delete;
+
+    char data[MESSAGE_LENGTH];
+    std::atomic<Message*> mNext;
+  };
+  // Creates a new MPSCQueue. Initially, the queue has a single sentinel node,
+  // pointed to by both mHead and mTail.
+  MPSCQueue()
+      // At construction, the initial message points to nullptr (it has no
+      // successor). It is a sentinel node, that does not contain meaningful
+      // data.
+      : mHead(new Message()), mTail(mHead.load(std::memory_order_relaxed)) {}
+
+  ~MPSCQueue() {
+    Message dummy;
+    while (this->Pop(dummy.data)) {
+    }
+    Message* front = mHead.load(std::memory_order_relaxed);
+    delete front;
+  }
 
-        char data[MESSAGE_LENGTH];
-        std::atomic<Message*> mNext;
-    };
-    // Creates a new MPSCQueue. Initially, the queue has a single sentinel node,
-    // pointed to by both mHead and mTail.
-    MPSCQueue()
-    // At construction, the initial message points to nullptr (it has no
-    // successor). It is a sentinel node, that does not contain meaningful
-    // data.
-    : mHead(new Message())
-    , mTail(mHead.load(std::memory_order_relaxed))
-    { }
-
-    ~MPSCQueue()
-    {
-        Message dummy;
-        while (this->Pop(dummy.data)) {}
-        Message* front = mHead.load(std::memory_order_relaxed);
-        delete front;
-    }
+  void Push(MPSCQueue<MESSAGE_LENGTH>::Message* aMessage) {
+    // The next two non-commented line are called A and B in this paragraph.
+    // Producer threads i, i-1, etc. are numbered in the order they reached
+    // A in time, thread i being the thread that has reached A first.
+    // Atomically, on line A the new `mHead` is set to be the node that was
+    // just allocated, with strong memory order. From now one, any thread
+    // that reaches A will see that the node just allocated is
+    // effectively the head of the list, and will make itself the new head
+    // of the list.
+    // In a bad case (when thread i executes A and then
+    // is not scheduled for a long time), it is possible that thread i-1 and
+    // subsequent threads create a seemingly disconnected set of nodes, but
+    // they all have the correct value for the next node to set as their
+    // mNext member on their respective stacks (in `prev`), and this is
+    // always correct. When the scheduler resumes, and line B is executed,
+    // the correct linkage is resumed.
+    // Before line B, since mNext for the node the was the last element of
+    // the queue still has an mNext of nullptr, Pop will not see the node
+    // added.
+    // For line A, it's critical to have strong ordering both ways (since
+    // it's going to possibly be read and write repeatidly by multiple
+    // threads)
+    // Line B can have weaker guarantees, it's only going to be written by a
+    // single thread, and we just need to ensure it's read properly by a
+    // single other one.
+    Message* prev = mHead.exchange(aMessage, std::memory_order_acq_rel);
+    prev->mNext.store(aMessage, std::memory_order_release);
+  }
 
-    void
-    Push(MPSCQueue<MESSAGE_LENGTH>::Message* aMessage)
-    {
-        // The next two non-commented line are called A and B in this paragraph.
-        // Producer threads i, i-1, etc. are numbered in the order they reached
-        // A in time, thread i being the thread that has reached A first.
-        // Atomically, on line A the new `mHead` is set to be the node that was
-        // just allocated, with strong memory order. From now one, any thread
-        // that reaches A will see that the node just allocated is
-        // effectively the head of the list, and will make itself the new head
-        // of the list.
-        // In a bad case (when thread i executes A and then
-        // is not scheduled for a long time), it is possible that thread i-1 and
-        // subsequent threads create a seemingly disconnected set of nodes, but
-        // they all have the correct value for the next node to set as their
-        // mNext member on their respective stacks (in `prev`), and this is
-        // always correct. When the scheduler resumes, and line B is executed,
-        // the correct linkage is resumed.
-        // Before line B, since mNext for the node the was the last element of
-        // the queue still has an mNext of nullptr, Pop will not see the node
-        // added.
-        // For line A, it's critical to have strong ordering both ways (since
-        // it's going to possibly be read and write repeatidly by multiple
-        // threads)
-        // Line B can have weaker guarantees, it's only going to be written by a
-        // single thread, and we just need to ensure it's read properly by a
-        // single other one.
-        Message* prev = mHead.exchange(aMessage, std::memory_order_acq_rel);
-        prev->mNext.store(aMessage, std::memory_order_release);
+  // Allocates a new node, copy aInput to the new memory location, and pushes
+  // it to the end of the list.
+  void Push(const char aInput[MESSAGE_LENGTH]) {
+    // Create a new message, and copy the messages passed on argument to the
+    // new memory location. We are not touching the queue right now. The
+    // successor for this new node is set to be nullptr.
+    Message* msg = new Message();
+    strncpy(msg->data, aInput, MESSAGE_LENGTH);
+
+    Push(msg);
+  }
+
+  // Copy the content of the first message of the queue to aOutput, and
+  // frees the message. Returns true if there was a message, in which case
+  // `aOutput` contains a valid value. If the queue was empty, returns false,
+  // in which case `aOutput` is left untouched.
+  bool Pop(char aOutput[MESSAGE_LENGTH]) {
+    // Similarly, in this paragraph, the two following lines are called A
+    // and B, and threads are called thread i, i-1, etc. in order of
+    // execution of line A.
+    // On line A, the first element of the queue is acquired. It is simply a
+    // sentinel node.
+    // On line B, we acquire the node that has the data we want. If B is
+    // null, then only the sentinel node was present in the queue, we can
+    // safely return false.
+    // mTail can be loaded with relaxed ordering, since it's not written nor
+    // read by any other thread (this queue is single consumer).
+    // mNext can be written to by one of the producer, so it's necessary to
+    // ensure those writes are seen, hence the stricter ordering.
+    Message* tail = mTail.load(std::memory_order_relaxed);
+    Message* next = tail->mNext.load(std::memory_order_acquire);
+
+    if (next == nullptr) {
+      return false;
     }
 
-    // Allocates a new node, copy aInput to the new memory location, and pushes
-    // it to the end of the list.
-    void
-    Push(const char aInput[MESSAGE_LENGTH])
-    {
-        // Create a new message, and copy the messages passed on argument to the
-        // new memory location. We are not touching the queue right now. The
-        // successor for this new node is set to be nullptr.
-        Message* msg = new Message();
-        strncpy(msg->data, aInput, MESSAGE_LENGTH);
-
-        Push(msg);
-    }
+    strncpy(aOutput, next->data, MESSAGE_LENGTH);
 
-    // Copy the content of the first message of the queue to aOutput, and
-    // frees the message. Returns true if there was a message, in which case
-    // `aOutput` contains a valid value. If the queue was empty, returns false,
-    // in which case `aOutput` is left untouched.
-    bool
-    Pop(char aOutput[MESSAGE_LENGTH])
-    {
-        // Similarly, in this paragraph, the two following lines are called A
-        // and B, and threads are called thread i, i-1, etc. in order of
-        // execution of line A.
-        // On line A, the first element of the queue is acquired. It is simply a
-        // sentinel node.
-        // On line B, we acquire the node that has the data we want. If B is
-        // null, then only the sentinel node was present in the queue, we can
-        // safely return false.
-        // mTail can be loaded with relaxed ordering, since it's not written nor
-        // read by any other thread (this queue is single consumer).
-        // mNext can be written to by one of the producer, so it's necessary to
-        // ensure those writes are seen, hence the stricter ordering.
-        Message* tail = mTail.load(std::memory_order_relaxed);
-        Message* next = tail->mNext.load(std::memory_order_acquire);
+    // Simply shift the queue one node further, so that the sentinel node is
+    // now pointing to the correct most ancient node. It contains stale data,
+    // but this data will never be read again.
+    // It's only necessary to ensure the previous load on this thread is not
+    // reordered past this line, so release ordering is sufficient here.
+    mTail.store(next, std::memory_order_release);
+
+    // This thread is now the only thing that points to `tail`, it can be
+    // safely deleted.
+    delete tail;
+
+    return true;
+  }
 
-        if (next == nullptr) {
-            return false;
-        }
-
-        strncpy(aOutput, next->data, MESSAGE_LENGTH);
+ private:
+  // An atomic pointer to the most recent message in the queue.
+  std::atomic<Message*> mHead;
+  // An atomic pointer to a sentinel node, that points to the oldest message
+  // in the queue.
+  std::atomic<Message*> mTail;
 
-        // Simply shift the queue one node further, so that the sentinel node is
-        // now pointing to the correct most ancient node. It contains stale data,
-        // but this data will never be read again.
-        // It's only necessary to ensure the previous load on this thread is not
-        // reordered past this line, so release ordering is sufficient here.
-        mTail.store(next, std::memory_order_release);
-
-        // This thread is now the only thing that points to `tail`, it can be
-        // safely deleted.
-        delete tail;
-
-        return true;
-    }
+  MPSCQueue(const MPSCQueue&) = delete;
+  void operator=(const MPSCQueue&) = delete;
 
-private:
-    // An atomic pointer to the most recent message in the queue.
-    std::atomic<Message*> mHead;
-    // An atomic pointer to a sentinel node, that points to the oldest message
-    // in the queue.
-    std::atomic<Message*> mTail;
+ public:
+  // The goal here is to make it easy on the allocator. We pack a pointer in the
+  // message struct, and we still want to do power of two allocations to
+  // minimize allocator slop. The allocation size are going to be constant, so
+  // the allocation is probably going to hit the thread local cache in jemalloc,
+  // making it cheap and, more importantly, lock-free enough.
+  static const size_t MESSAGE_PADDING = sizeof(Message::mNext);
 
-    MPSCQueue(const MPSCQueue&) = delete;
-    void operator=(const MPSCQueue&) = delete;
-public:
-    // The goal here is to make it easy on the allocator. We pack a pointer in the
-    // message struct, and we still want to do power of two allocations to
-    // minimize allocator slop. The allocation size are going to be constant, so
-    // the allocation is probably going to hit the thread local cache in jemalloc,
-    // making it cheap and, more importantly, lock-free enough.
-    static const size_t MESSAGE_PADDING = sizeof(Message::mNext);
-private:
-    static_assert(IsPowerOfTwo(MESSAGE_LENGTH + MESSAGE_PADDING),
-                  "MPSCQueue internal allocations must have a size that is a"
-                  "power of two ");
+ private:
+  static_assert(IsPowerOfTwo(MESSAGE_LENGTH + MESSAGE_PADDING),
+                "MPSCQueue internal allocations must have a size that is a"
+                "power of two ");
 };
-} // end namespace detail
+}  // end namespace detail
 
 // This class implements a lock-free asynchronous logger, that outputs to
 // MOZ_LOG.
 // Any thread can use this logger without external synchronization and without
 // being blocked. This log is suitable for use in real-time audio threads.
 // Log formatting is best done externally, this class implements the output
 // mechanism only.
 // This class uses a thread internally, and must be started and stopped
 // manually.
 // If logging is disabled, all the calls are no-op.
-class AsyncLogger
-{
-public:
-  static const uint32_t MAX_MESSAGE_LENGTH = 512 - detail::MPSCQueue<sizeof(void*)>::MESSAGE_PADDING;
+class AsyncLogger {
+ public:
+  static const uint32_t MAX_MESSAGE_LENGTH =
+      512 - detail::MPSCQueue<sizeof(void*)>::MESSAGE_PADDING;
 
   // aLogModuleName is the name of the MOZ_LOG module.
   explicit AsyncLogger(const char* aLogModuleName)
-  : mThread(nullptr)
-  , mLogModule(aLogModuleName)
-  , mRunning(false)
-  { }
+      : mThread(nullptr), mLogModule(aLogModuleName), mRunning(false) {}
 
-  ~AsyncLogger()
-  {
+  ~AsyncLogger() {
     if (Enabled()) {
       Stop();
     }
   }
 
-  void Start()
-  {
+  void Start() {
     MOZ_ASSERT(!mRunning, "Double calls to AsyncLogger::Start");
     if (Enabled()) {
       mRunning = true;
       Run();
     }
   }
 
-  void Stop()
-  {
+  void Stop() {
     if (Enabled()) {
       if (mRunning) {
         mRunning = false;
         mThread->join();
       }
     } else {
       MOZ_ASSERT(!mRunning && !mThread);
     }
   }
 
-  void Log(const char* format, ...) MOZ_FORMAT_PRINTF(2,3)
-  {
+  void Log(const char* format, ...) MOZ_FORMAT_PRINTF(2, 3) {
     if (Enabled()) {
       auto* msg = new detail::MPSCQueue<MAX_MESSAGE_LENGTH>::Message();
       va_list args;
       va_start(args, format);
       VsprintfLiteral(msg->data, format, args);
       va_end(args);
       mMessageQueue.Push(msg);
     }
   }
 
-  bool Enabled()
-  {
+  bool Enabled() {
     return MOZ_LOG_TEST(mLogModule, mozilla::LogLevel::Verbose);
   }
 
-private:
-  void Run()
-  {
+ private:
+  void Run() {
     MOZ_ASSERT(Enabled());
     mThread.reset(new std::thread([this]() {
       while (mRunning) {
         char message[MAX_MESSAGE_LENGTH];
         while (mMessageQueue.Pop(message) && mRunning) {
           MOZ_LOG(mLogModule, mozilla::LogLevel::Verbose, ("%s", message));
         }
         Sleep();
@@ -259,11 +240,11 @@ private:
   void Sleep() { std::this_thread::sleep_for(std::chrono::milliseconds(10)); }
 
   std::unique_ptr<std::thread> mThread;
   mozilla::LazyLogModule mLogModule;
   detail::MPSCQueue<MAX_MESSAGE_LENGTH> mMessageQueue;
   std::atomic<bool> mRunning;
 };
 
-} // end namespace mozilla
+}  // end namespace mozilla
 
-#endif // mozilla_dom_AsyncLogger_h
+#endif  // mozilla_dom_AsyncLogger_h

--- a/dom/media/AudioBufferUtils.h
+++ b/dom/media/AudioBufferUtils.h
@@ -24,39 +24,34 @@ static inline uint32_t SamplesToFrames(u
   return aSamples / aChannels;
 }
 
 /**
  * Class that gets a buffer pointer from an audio callback and provides a safe
  * interface to manipulate this buffer, and to ensure we are not missing frames
  * by the end of the callback.
  */
-template<typename T>
-class AudioCallbackBufferWrapper
-{
-public:
+template <typename T>
+class AudioCallbackBufferWrapper {
+ public:
   AudioCallbackBufferWrapper()
-    : mBuffer(nullptr)
-    , mSamples(0)
-    , mSampleWriteOffset(1)
-    , mChannels(0)
-  {}
+      : mBuffer(nullptr), mSamples(0), mSampleWriteOffset(1), mChannels(0) {}
 
   explicit AudioCallbackBufferWrapper(uint32_t aChannels)
-    : mBuffer(nullptr)
-    , mSamples(0)
-    , mSampleWriteOffset(1)
-    , mChannels(aChannels)
+      : mBuffer(nullptr),
+        mSamples(0),
+        mSampleWriteOffset(1),
+        mChannels(aChannels)
 
   {
     MOZ_ASSERT(aChannels);
   }
 
-  AudioCallbackBufferWrapper& operator=(const AudioCallbackBufferWrapper& aOther)
-  {
+  AudioCallbackBufferWrapper& operator=(
+      const AudioCallbackBufferWrapper& aOther) {
     MOZ_ASSERT(!aOther.mBuffer,
                "Don't use this ctor after AudioCallbackDriver::Init");
     MOZ_ASSERT(aOther.mSamples == 0,
                "Don't use this ctor after AudioCallbackDriver::Init");
     MOZ_ASSERT(aOther.mSampleWriteOffset == 1,
                "Don't use this ctor after AudioCallbackDriver::Init");
     MOZ_ASSERT(aOther.mChannels != 0);
 
@@ -68,33 +63,32 @@ public:
     return *this;
   }
 
   /**
    * Set the buffer in this wrapper. This is to be called at the beginning of
    * the callback.
    */
   void SetBuffer(T* aBuffer, uint32_t aFrames) {
-    MOZ_ASSERT(!mBuffer && !mSamples,
-        "SetBuffer called twice.");
+    MOZ_ASSERT(!mBuffer && !mSamples, "SetBuffer called twice.");
     mBuffer = aBuffer;
     mSamples = FramesToSamples(mChannels, aFrames);
     mSampleWriteOffset = 0;
   }
 
   /**
    * Write some frames to the internal buffer. Free space in the buffer should
    * be check prior to calling this.
    */
   void WriteFrames(T* aBuffer, uint32_t aFrames) {
     MOZ_ASSERT(aFrames <= Available(),
-        "Writing more that we can in the audio buffer.");
+               "Writing more that we can in the audio buffer.");
 
-    PodCopy(mBuffer + mSampleWriteOffset, aBuffer, FramesToSamples(mChannels,
-                                                                   aFrames));
+    PodCopy(mBuffer + mSampleWriteOffset, aBuffer,
+            FramesToSamples(mChannels, aFrames));
     mSampleWriteOffset += FramesToSamples(mChannels, aFrames);
   }
 
   /**
    * Number of frames that can be written to the buffer.
    */
   uint32_t Available() {
     return SamplesToFrames(mChannels, mSamples - mSampleWriteOffset);
@@ -102,36 +96,37 @@ public:
 
   /**
    * Check that the buffer is completly filled, and reset internal state so this
    * instance can be reused.
    */
   void BufferFilled() {
     // It's okay to have exactly zero samples here, it can happen we have an
     // audio callback driver because of a hint on MSG creation, but the
-    // AudioOutputStream has not been created yet, or if all the streams have finished
-    // but we're still running.
-    // Note: it's also ok if we had data in the scratch buffer - and we usually do - and
-    // all the streams were ended (no mixer callback occured).
+    // AudioOutputStream has not been created yet, or if all the streams have
+    // finished but we're still running. Note: it's also ok if we had data in
+    // the scratch buffer - and we usually do - and all the streams were ended
+    // (no mixer callback occured).
     // XXX Remove this warning, or find a way to avoid it if the mixer callback
     // isn't called.
     NS_WARNING_ASSERTION(
-      Available() == 0 || mSampleWriteOffset == 0,
-      "Audio Buffer is not full by the end of the callback.");
+        Available() == 0 || mSampleWriteOffset == 0,
+        "Audio Buffer is not full by the end of the callback.");
     // Make sure the data returned is always set and not random!
     if (Available()) {
-      PodZero(mBuffer + mSampleWriteOffset, FramesToSamples(mChannels, Available()));
+      PodZero(mBuffer + mSampleWriteOffset,
+              FramesToSamples(mChannels, Available()));
     }
     MOZ_ASSERT(mSamples, "Buffer not set.");
     mSamples = 0;
     mSampleWriteOffset = 0;
     mBuffer = nullptr;
   }
 
-private:
+ private:
   /* This is not an owned pointer, but the pointer passed to use via the audio
    * callback. */
   T* mBuffer;
   /* The number of samples of this audio buffer. */
   uint32_t mSamples;
   /* The position at which new samples should be written. We want to return to
    * the audio callback iff this is equal to mSamples. */
   uint32_t mSampleWriteOffset;
@@ -139,91 +134,86 @@ private:
 };
 
 /**
  * This is a class that interfaces with the AudioCallbackBufferWrapper, and is
  * responsible for storing the excess of data produced by the MediaStreamGraph
  * because of different rounding constraints, to be used the next time the audio
  * backend calls back.
  */
-template<typename T, uint32_t BLOCK_SIZE>
-class SpillBuffer
-{
-public:
-  SpillBuffer()
-    : mBuffer(nullptr)
-    , mPosition(0)
-    , mChannels(0)
-  {}
+template <typename T, uint32_t BLOCK_SIZE>
+class SpillBuffer {
+ public:
+  SpillBuffer() : mBuffer(nullptr), mPosition(0), mChannels(0) {}
 
   explicit SpillBuffer(uint32_t aChannels)
-  : mPosition(0)
-  , mChannels(aChannels)
-  {
+      : mPosition(0), mChannels(aChannels) {
     MOZ_ASSERT(aChannels);
     mBuffer = MakeUnique<T[]>(BLOCK_SIZE * mChannels);
     PodZero(mBuffer.get(), BLOCK_SIZE * mChannels);
   }
 
-  SpillBuffer& operator=(SpillBuffer& aOther)
-  {
+  SpillBuffer& operator=(SpillBuffer& aOther) {
     MOZ_ASSERT(aOther.mPosition == 0,
-        "Don't use this ctor after AudioCallbackDriver::Init");
+               "Don't use this ctor after AudioCallbackDriver::Init");
     MOZ_ASSERT(aOther.mChannels != 0);
     MOZ_ASSERT(aOther.mBuffer);
 
     mPosition = aOther.mPosition;
     mChannels = aOther.mChannels;
     mBuffer = std::move(aOther.mBuffer);
 
     return *this;
   }
 
-  SpillBuffer& operator=(SpillBuffer&& aOther)
-  {
+  SpillBuffer& operator=(SpillBuffer&& aOther) {
     return this->operator=(aOther);
   }
 
   /* Empty the spill buffer into the buffer of the audio callback. This returns
    * the number of frames written. */
   uint32_t Empty(AudioCallbackBufferWrapper<T>& aBuffer) {
-    uint32_t framesToWrite = std::min(aBuffer.Available(),
-                                      SamplesToFrames(mChannels, mPosition));
+    uint32_t framesToWrite =
+        std::min(aBuffer.Available(), SamplesToFrames(mChannels, mPosition));
 
     aBuffer.WriteFrames(mBuffer.get(), framesToWrite);
 
     mPosition -= FramesToSamples(mChannels, framesToWrite);
-    // If we didn't empty the spill buffer for some reason, shift the remaining data down
+    // If we didn't empty the spill buffer for some reason, shift the remaining
+    // data down
     if (mPosition > 0) {
-      MOZ_ASSERT(FramesToSamples(mChannels, framesToWrite) + mPosition <= BLOCK_SIZE * mChannels);
-      PodMove(mBuffer.get(), mBuffer.get() + FramesToSamples(mChannels, framesToWrite),
+      MOZ_ASSERT(FramesToSamples(mChannels, framesToWrite) + mPosition <=
+                 BLOCK_SIZE * mChannels);
+      PodMove(mBuffer.get(),
+              mBuffer.get() + FramesToSamples(mChannels, framesToWrite),
               mPosition);
     }
 
     return framesToWrite;
   }
   /* Fill the spill buffer from aInput, containing aFrames frames, return the
    * number of frames written to the spill buffer */
   uint32_t Fill(T* aInput, uint32_t aFrames) {
-    uint32_t framesToWrite = std::min(aFrames,
-                                      BLOCK_SIZE - SamplesToFrames(mChannels,
-                                                                   mPosition));
+    uint32_t framesToWrite =
+        std::min(aFrames, BLOCK_SIZE - SamplesToFrames(mChannels, mPosition));
 
-    MOZ_ASSERT(FramesToSamples(mChannels, framesToWrite) + mPosition <= BLOCK_SIZE * mChannels);
-    PodCopy(mBuffer.get() + mPosition, aInput, FramesToSamples(mChannels,
-                                                         framesToWrite));
+    MOZ_ASSERT(FramesToSamples(mChannels, framesToWrite) + mPosition <=
+               BLOCK_SIZE * mChannels);
+    PodCopy(mBuffer.get() + mPosition, aInput,
+            FramesToSamples(mChannels, framesToWrite));
 
     mPosition += FramesToSamples(mChannels, framesToWrite);
 
     return framesToWrite;
   }
-private:
+
+ private:
   /* The spilled data. */
   UniquePtr<T[]> mBuffer;
   /* The current write position, in samples, in the buffer when filling, or the
    * amount of buffer filled when emptying. */
   uint32_t mPosition;
   uint32_t mChannels;
 };
 
-} // namespace mozilla
+}  // namespace mozilla
 
-#endif // MOZILLA_SCRATCHBUFFER_H_
+#endif  // MOZILLA_SCRATCHBUFFER_H_

--- a/dom/media/AudioCaptureStream.cpp
+++ b/dom/media/AudioCaptureStream.cpp
@@ -19,75 +19,65 @@
 #include "webaudio/MediaStreamAudioDestinationNode.h"
 #include <algorithm>
 #include "DOMMediaStream.h"
 
 using namespace mozilla::layers;
 using namespace mozilla::dom;
 using namespace mozilla::gfx;
 
-namespace mozilla
-{
+namespace mozilla {
 
 // We are mixing to mono until PeerConnection can accept stereo
 static const uint32_t MONO = 1;
 
 AudioCaptureStream::AudioCaptureStream(TrackID aTrackId)
-  : ProcessedMediaStream()
-  , mTrackId(aTrackId)
-  , mStarted(false)
-  , mTrackCreated(false)
-{
+    : ProcessedMediaStream(),
+      mTrackId(aTrackId),
+      mStarted(false),
+      mTrackCreated(false) {
   MOZ_ASSERT(NS_IsMainThread());
   MOZ_COUNT_CTOR(AudioCaptureStream);
   mMixer.AddCallback(this);
 }
 
-AudioCaptureStream::~AudioCaptureStream()
-{
+AudioCaptureStream::~AudioCaptureStream() {
   MOZ_COUNT_DTOR(AudioCaptureStream);
   mMixer.RemoveCallback(this);
 }
 
-void
-AudioCaptureStream::Start()
-{
+void AudioCaptureStream::Start() {
   class Message : public ControlMessage {
-  public:
+   public:
     explicit Message(AudioCaptureStream* aStream)
-      : ControlMessage(aStream), mStream(aStream) {}
+        : ControlMessage(aStream), mStream(aStream) {}
 
-    virtual void Run()
-    {
-      mStream->mStarted = true;
-    }
+    virtual void Run() { mStream->mStarted = true; }
 
-  protected:
+   protected:
     AudioCaptureStream* mStream;
   };
   GraphImpl()->AppendMessage(MakeUnique<Message>(this));
 }
 
-void
-AudioCaptureStream::ProcessInput(GraphTime aFrom, GraphTime aTo,
-                                 uint32_t aFlags)
-{
+void AudioCaptureStream::ProcessInput(GraphTime aFrom, GraphTime aTo,
+                                      uint32_t aFlags) {
   if (!mStarted) {
     return;
   }
 
   uint32_t inputCount = mInputs.Length();
   StreamTracks::Track* track = EnsureTrack(mTrackId);
   // Notify the DOM everything is in order.
   if (!mTrackCreated) {
     for (uint32_t i = 0; i < mListeners.Length(); i++) {
       MediaStreamListener* l = mListeners[i];
       AudioSegment tmp;
-      l->NotifyQueuedTrackChanges(
-        Graph(), mTrackId, 0, TrackEventCommand::TRACK_EVENT_CREATED, tmp);
+      l->NotifyQueuedTrackChanges(Graph(), mTrackId, 0,
+                                  TrackEventCommand::TRACK_EVENT_CREATED, tmp);
       l->NotifyFinishedTrackCreation(Graph());
     }
     mTrackCreated = true;
   }
 
   if (IsFinishedOnGraphThread()) {
     return;
   }
@@ -132,40 +122,40 @@ AudioCaptureStream::ProcessInput(GraphTi
     // This calls MixerCallback below
     mMixer.FinishMixing();
   }
 
   // Regardless of the status of the input tracks, we go foward.
   mTracks.AdvanceKnownTracksTime(GraphTimeToStreamTimeWithBlocking((aTo)));
 }
 
-void
-AudioCaptureStream::MixerCallback(AudioDataValue* aMixedBuffer,
-                                  AudioSampleFormat aFormat, uint32_t aChannels,
-                                  uint32_t aFrames, uint32_t aSampleRate)
-{
+void AudioCaptureStream::MixerCallback(AudioDataValue* aMixedBuffer,
+                                       AudioSampleFormat aFormat,
+                                       uint32_t aChannels, uint32_t aFrames,
+                                       uint32_t aSampleRate) {
   AutoTArray<nsTArray<AudioDataValue>, MONO> output;
   AutoTArray<const AudioDataValue*, MONO> bufferPtrs;
   output.SetLength(MONO);
   bufferPtrs.SetLength(MONO);
 
   uint32_t written = 0;
   // We need to copy here, because the mixer will reuse the storage, we should
   // not hold onto it. Buffers are in planar format.
   for (uint32_t channel = 0; channel < aChannels; channel++) {
     AudioDataValue* out = output[channel].AppendElements(aFrames);
     PodCopy(out, aMixedBuffer + written, aFrames);
     bufferPtrs[channel] = out;
     written += aFrames;
   }
   AudioChunk chunk;
-  chunk.mBuffer = new mozilla::SharedChannelArrayBuffer<AudioDataValue>(&output);
+  chunk.mBuffer =
+      new mozilla::SharedChannelArrayBuffer<AudioDataValue>(&output);
   chunk.mDuration = aFrames;
   chunk.mBufferFormat = aFormat;
   chunk.mChannelData.SetLength(MONO);
   for (uint32_t channel = 0; channel < aChannels; channel++) {
     chunk.mChannelData[channel] = bufferPtrs[channel];
   }
 
   // Now we have mixed data, simply append it to out track.
   EnsureTrack(mTrackId)->Get<AudioSegment>()->AppendAndConsumeChunk(&chunk);
 }
-}
+}  // namespace mozilla

--- a/dom/media/AudioCaptureStream.h
+++ b/dom/media/AudioCaptureStream.h
@@ -6,40 +6,38 @@
 #ifndef MOZILLA_AUDIOCAPTURESTREAM_H_
 #define MOZILLA_AUDIOCAPTURESTREAM_H_
 
 #include "MediaStreamGraph.h"
 #include "AudioMixer.h"
 #include "StreamTracks.h"
 #include <algorithm>
 
-namespace mozilla
-{
+namespace mozilla {
 
 class AbstractThread;
 class DOMMediaStream;
 
 /**
  * See MediaStreamGraph::CreateAudioCaptureStream.
  */
 class AudioCaptureStream : public ProcessedMediaStream,
-                           public MixerCallbackReceiver
-{
-public:
+                           public MixerCallbackReceiver {
+ public:
   explicit AudioCaptureStream(TrackID aTrackId);
   virtual ~AudioCaptureStream();
 
   void Start();
 
   void ProcessInput(GraphTime aFrom, GraphTime aTo, uint32_t aFlags) override;
 
-protected:
+ protected:
   void MixerCallback(AudioDataValue* aMixedBuffer, AudioSampleFormat aFormat,
                      uint32_t aChannels, uint32_t aFrames,
                      uint32_t aSampleRate) override;
   AudioMixer mMixer;
   TrackID mTrackId;
   bool mStarted;
   bool mTrackCreated;
 };
-}
+}  // namespace mozilla
 
 #endif /* MOZILLA_AUDIOCAPTURESTREAM_H_ */

--- a/dom/media/AudioChannelFormat.cpp
+++ b/dom/media/AudioChannelFormat.cpp
@@ -4,15 +4,13 @@
  * You can obtain one at http://mozilla.org/MPL/2.0/. */
 
 #include "AudioChannelFormat.h"
 
 #include <algorithm>
 
 namespace mozilla {
 
-uint32_t
-GetAudioChannelsSuperset(uint32_t aChannels1, uint32_t aChannels2)
-{
+uint32_t GetAudioChannelsSuperset(uint32_t aChannels1, uint32_t aChannels2) {
   return std::max(aChannels1, aChannels2);
 }
 
-} // namespace mozilla
+}  // namespace mozilla

--- a/dom/media/AudioChannelFormat.h
+++ b/dom/media/AudioChannelFormat.h
@@ -43,25 +43,24 @@ enum {
 const uint32_t CUSTOM_CHANNEL_LAYOUTS = 6;
 
 // This is defined by some Windows SDK header.
 #undef IGNORE
 
 const int IGNORE = CUSTOM_CHANNEL_LAYOUTS;
 const float IGNORE_F = 0.0f;
 
-const int gMixingMatrixIndexByChannels[CUSTOM_CHANNEL_LAYOUTS - 1] =
-  { 0, 5, 9, 12, 14 };
+const int gMixingMatrixIndexByChannels[CUSTOM_CHANNEL_LAYOUTS - 1] = {0, 5, 9,
+                                                                      12, 14};
 
 /**
  * Return a channel count whose channel layout includes all the channels from
  * aChannels1 and aChannels2.
  */
-uint32_t
-GetAudioChannelsSuperset(uint32_t aChannels1, uint32_t aChannels2);
+uint32_t GetAudioChannelsSuperset(uint32_t aChannels1, uint32_t aChannels2);
 
 /**
  * DownMixMatrix represents a conversion matrix efficiently by exploiting the
  * fact that each input channel contributes to at most one output channel,
  * except possibly for the C input channel in layouts that have one. Also,
  * every input channel is multiplied by the same coefficient for every output
  * channel it contributes to.
  */
@@ -72,160 +71,165 @@ struct DownMixMatrix {
   // after multiplying by mInputCoefficient[c].
   uint8_t mInputDestination[CUSTOM_CHANNEL_LAYOUTS];
   // If not IGNORE, then the C channel is copied to this output channel after
   // multiplying by its coefficient.
   uint8_t mCExtraDestination;
   float mInputCoefficient[CUSTOM_CHANNEL_LAYOUTS];
 };
 
-static const DownMixMatrix
-gDownMixMatrices[CUSTOM_CHANNEL_LAYOUTS*(CUSTOM_CHANNEL_LAYOUTS - 1)/2] =
-{
-  // Downmixes to mono
-  { { 0, 0 }, IGNORE, { 0.5f, 0.5f } },
-  { { 0, IGNORE, IGNORE }, IGNORE, { 1.0f, IGNORE_F, IGNORE_F } },
-  { { 0, 0, 0, 0 }, IGNORE, { 0.25f, 0.25f, 0.25f, 0.25f } },
-  { { 0, IGNORE, IGNORE, IGNORE, IGNORE }, IGNORE, { 1.0f, IGNORE_F, IGNORE_F, IGNORE_F, IGNORE_F } },
-  { { 0, 0, 0, IGNORE, 0, 0 }, IGNORE, { SQRT_ONE_HALF, SQRT_ONE_HALF, 1.0f, IGNORE_F, 0.5f, 0.5f } },
-  // Downmixes to stereo
-  { { 0, 1, IGNORE }, IGNORE, { 1.0f, 1.0f, IGNORE_F } },
-  { { 0, 1, 0, 1 }, IGNORE, { 0.5f, 0.5f, 0.5f, 0.5f } },
-  { { 0, 1, IGNORE, IGNORE, IGNORE }, IGNORE, { 1.0f, 1.0f, IGNORE_F, IGNORE_F, IGNORE_F } },
-  { { 0, 1, 0, IGNORE, 0, 1 }, 1, { 1.0f, 1.0f, SQRT_ONE_HALF, IGNORE_F, SQRT_ONE_HALF, SQRT_ONE_HALF } },
-  // Downmixes to 3-channel
-  { { 0, 1, 2, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, IGNORE_F } },
-  { { 0, 1, 2, IGNORE, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, IGNORE_F, IGNORE_F } },
-  { { 0, 1, 2, IGNORE, IGNORE, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, IGNORE_F, IGNORE_F, IGNORE_F } },
-  // Downmixes to quad
-  { { 0, 1, 2, 3, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, 1.0f, IGNORE_F } },
-  { { 0, 1, 0, IGNORE, 2, 3 }, 1, { 1.0f, 1.0f, SQRT_ONE_HALF, IGNORE_F, 1.0f, 1.0f } },
-  // Downmixes to 5-channel
-  { { 0, 1, 2, 3, 4, IGNORE }, IGNORE, { 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, IGNORE_F } }
-};
+static const DownMixMatrix gDownMixMatrices[CUSTOM_CHANNEL_LAYOUTS *
+                                            (CUSTOM_CHANNEL_LAYOUTS - 1) /
+                                            2] = {
+    // Downmixes to mono
+    {{0, 0}, IGNORE, {0.5f, 0.5f}},
+    {{0, IGNORE, IGNORE}, IGNORE, {1.0f, IGNORE_F, IGNORE_F}},
+    {{0, 0, 0, 0}, IGNORE, {0.25f, 0.25f, 0.25f, 0.25f}},
+    {{0, IGNORE, IGNORE, IGNORE, IGNORE},
+     IGNORE,
+     {1.0f, IGNORE_F, IGNORE_F, IGNORE_F, IGNORE_F}},
+    {{0, 0, 0, IGNORE, 0, 0},
+     IGNORE,
+     {SQRT_ONE_HALF, SQRT_ONE_HALF, 1.0f, IGNORE_F, 0.5f, 0.5f}},
+    // Downmixes to stereo
+    {{0, 1, IGNORE}, IGNORE, {1.0f, 1.0f, IGNORE_F}},
+    {{0, 1, 0, 1}, IGNORE, {0.5f, 0.5f, 0.5f, 0.5f}},
+    {{0, 1, IGNORE, IGNORE, IGNORE},
+     IGNORE,
+     {1.0f, 1.0f, IGNORE_F, IGNORE_F, IGNORE_F}},
+    {{0, 1, 0, IGNORE, 0, 1},
+     1,
+     {1.0f, 1.0f, SQRT_ONE_HALF, IGNORE_F, SQRT_ONE_HALF, SQRT_ONE_HALF}},
+    // Downmixes to 3-channel
+    {{0, 1, 2, IGNORE}, IGNORE, {1.0f, 1.0f, 1.0f, IGNORE_F}},
+    {{0, 1, 2, IGNORE, IGNORE}, IGNORE, {1.0f, 1.0f, 1.0f, IGNORE_F, IGNORE_F}},
+    {{0, 1, 2, IGNORE, IGNORE, IGNORE},
+     IGNORE,
+     {1.0f, 1.0f, 1.0f, IGNORE_F, IGNORE_F, IGNORE_F}},
+    // Downmixes to quad
+    {{0, 1, 2, 3, IGNORE}, IGNORE, {1.0f, 1.0f, 1.0f, 1.0f, IGNORE_F}},
+    {{0, 1, 0, IGNORE, 2, 3},
+     1,
+     {1.0f, 1.0f, SQRT_ONE_HALF, IGNORE_F, 1.0f, 1.0f}},
+    // Downmixes to 5-channel
+    {{0, 1, 2, 3, 4, IGNORE},
+     IGNORE,
+     {1.0f, 1.0f, 1.0f, 1.0f, 1.0f, IGNORE_F}}};
 
 /**
  * Given an array of input channels, downmix to aOutputChannelCount, and copy
  * the results to the channel buffers in aOutputChannels.  Don't call this with
  * input count <= output count.
  */
-template<typename T>
+template <typename T>
 void AudioChannelsDownMix(const nsTArray<const T*>& aChannelArray,
-                          T** aOutputChannels,
-                          uint32_t aOutputChannelCount,
-                          uint32_t aDuration)
-{
+                          T** aOutputChannels, uint32_t aOutputChannelCount,
+                          uint32_t aDuration) {
   uint32_t inputChannelCount = aChannelArray.Length();
   const T* const* inputChannels = aChannelArray.Elements();
   NS_ASSERTION(inputChannelCount > aOutputChannelCount, "Nothing to do");
 
   if (inputChannelCount > 6) {
     // Just drop the unknown channels.
     for (uint32_t o = 0; o < aOutputChannelCount; ++o) {
       PodCopy(aOutputChannels[o], inputChannels[o], aDuration);
     }
     return;
   }
 
   // Ignore unknown channels, they're just dropped.
   inputChannelCount = std::min<uint32_t>(6, inputChannelCount);
 
-  const DownMixMatrix& m = gDownMixMatrices[
-    gMixingMatrixIndexByChannels[aOutputChannelCount - 1] +
-    inputChannelCount - aOutputChannelCount - 1];
+  const DownMixMatrix& m =
+      gDownMixMatrices[gMixingMatrixIndexByChannels[aOutputChannelCount - 1] +
+                       inputChannelCount - aOutputChannelCount - 1];
 
   // This is slow, but general. We can define custom code for special
   // cases later.
   for (uint32_t s = 0; s < aDuration; ++s) {
     // Reserve an extra junk channel at the end for the cases where we
     // want an input channel to contribute to nothing
     T outputChannels[CUSTOM_CHANNEL_LAYOUTS + 1] = {0};
     for (uint32_t c = 0; c < inputChannelCount; ++c) {
       outputChannels[m.mInputDestination[c]] +=
-        m.mInputCoefficient[c]*(static_cast<const T*>(inputChannels[c]))[s];
+          m.mInputCoefficient[c] * (static_cast<const T*>(inputChannels[c]))[s];
     }
     // Utilize the fact that in every layout, C is the third channel.
     if (m.mCExtraDestination != IGNORE) {
       outputChannels[m.mCExtraDestination] +=
-        m.mInputCoefficient[SURROUND_C]*(static_cast<const T*>(inputChannels[SURROUND_C]))[s];
+          m.mInputCoefficient[SURROUND_C] *
+          (static_cast<const T*>(inputChannels[SURROUND_C]))[s];
     }
 
     for (uint32_t c = 0; c < aOutputChannelCount; ++c) {
       aOutputChannels[c][s] = outputChannels[c];
     }
   }
 }
 
 /**
  * UpMixMatrix represents a conversion matrix by exploiting the fact that
  * each output channel comes from at most one input channel.
  */
 struct UpMixMatrix {
   uint8_t mInputDestination[CUSTOM_CHANNEL_LAYOUTS];
 };
 
-static const UpMixMatrix
-gUpMixMatrices[CUSTOM_CHANNEL_LAYOUTS*(CUSTOM_CHANNEL_LAYOUTS - 1)/2] =
-{
-  // Upmixes from mono
-  { { 0, 0 } },
-  { { 0, IGNORE, IGNORE } },
-  { { 0, 0, IGNORE, IGNORE } },
-  { { 0, IGNORE, IGNORE, IGNORE, IGNORE } },
-  { { IGNORE, IGNORE, 0, IGNORE, IGNORE, IGNORE } },
-  // Upmixes from stereo
-  { { 0, 1, IGNORE } },
-  { { 0, 1, IGNORE, IGNORE } },
-  { { 0, 1, IGNORE, IGNORE, IGNORE } },
-  { { 0, 1, IGNORE, IGNORE, IGNORE, IGNORE } },
-  // Upmixes from 3-channel
-  { { 0, 1, 2, IGNORE } },
-  { { 0, 1, 2, IGNORE, IGNORE } },
-  { { 0, 1, 2, IGNORE, IGNORE, IGNORE } },
-  // Upmixes from quad
-  { { 0, 1, 2, 3, IGNORE } },
-  { { 0, 1, IGNORE, IGNORE, 2, 3 } },
-  // Upmixes from 5-channel
-  { { 0, 1, 2, 3, 4, IGNORE } }
-};
-
+static const UpMixMatrix gUpMixMatrices[CUSTOM_CHANNEL_LAYOUTS *
+                                        (CUSTOM_CHANNEL_LAYOUTS - 1) / 2] = {
+    // Upmixes from mono
+    {{0, 0}},
+    {{0, IGNORE, IGNORE}},
+    {{0, 0, IGNORE, IGNORE}},
+    {{0, IGNORE, IGNORE, IGNORE, IGNORE}},
+    {{IGNORE, IGNORE, 0, IGNORE, IGNORE, IGNORE}},
+    // Upmixes from stereo
+    {{0, 1, IGNORE}},
+    {{0, 1, IGNORE, IGNORE}},
+    {{0, 1, IGNORE, IGNORE, IGNORE}},
+    {{0, 1, IGNORE, IGNORE, IGNORE, IGNORE}},
+    // Upmixes from 3-channel
+    {{0, 1, 2, IGNORE}},
+    {{0, 1, 2, IGNORE, IGNORE}},
+    {{0, 1, 2, IGNORE, IGNORE, IGNORE}},
+    // Upmixes from quad
+    {{0, 1, 2, 3, IGNORE}},
+    {{0, 1, IGNORE, IGNORE, 2, 3}},
+    // Upmixes from 5-channel
+    {{0, 1, 2, 3, 4, IGNORE}}};
 
 /**
  * Given an array of input channel data, and an output channel count,
  * replaces the array with an array of upmixed channels.
  * This shuffles the array and may set some channel buffers to aZeroChannel.
  * Don't call this with input count >= output count.
  * This may return *more* channels than requested. In that case, downmixing
  * is required to to get to aOutputChannelCount. (This is how we handle
  * odd cases like 3 -> 4 upmixing.)
  * If aChannelArray.Length() was the input to one of a series of
  * GetAudioChannelsSuperset calls resulting in aOutputChannelCount,
  * no downmixing will be required.
  */
-template<typename T>
-void
-AudioChannelsUpMix(nsTArray<const T*>* aChannelArray,
-                   uint32_t aOutputChannelCount,
-                   const T* aZeroChannel)
-{
+template <typename T>
+void AudioChannelsUpMix(nsTArray<const T*>* aChannelArray,
+                        uint32_t aOutputChannelCount, const T* aZeroChannel) {
   uint32_t inputChannelCount = aChannelArray->Length();
   uint32_t outputChannelCount =
-    GetAudioChannelsSuperset(aOutputChannelCount, inputChannelCount);
-  NS_ASSERTION(outputChannelCount > inputChannelCount,
-               "No up-mix needed");
+      GetAudioChannelsSuperset(aOutputChannelCount, inputChannelCount);
+  NS_ASSERTION(outputChannelCount > inputChannelCount, "No up-mix needed");
   MOZ_ASSERT(inputChannelCount > 0, "Bad number of channels");
   MOZ_ASSERT(outputChannelCount > 0, "Bad number of channels");
 
   aChannelArray->SetLength(outputChannelCount);
 
   if (inputChannelCount < CUSTOM_CHANNEL_LAYOUTS &&
       outputChannelCount <= CUSTOM_CHANNEL_LAYOUTS) {
-    const UpMixMatrix& m = gUpMixMatrices[
-      gMixingMatrixIndexByChannels[inputChannelCount - 1] +
-      outputChannelCount - inputChannelCount - 1];
+    const UpMixMatrix& m =
+        gUpMixMatrices[gMixingMatrixIndexByChannels[inputChannelCount - 1] +
+                       outputChannelCount - inputChannelCount - 1];
 
     const T* outputChannels[CUSTOM_CHANNEL_LAYOUTS];
 
     for (uint32_t i = 0; i < outputChannelCount; ++i) {
       uint8_t channelIndex = m.mInputDestination[i];
       if (channelIndex == IGNORE) {
         outputChannels[i] = aZeroChannel;
       } else {
@@ -238,11 +242,11 @@ AudioChannelsUpMix(nsTArray<const T*>* a
     return;
   }
 
   for (uint32_t i = inputChannelCount; i < outputChannelCount; ++i) {
     aChannelArray->ElementAt(i) = aZeroChannel;
   }
 }
 
-} // namespace mozilla
+}  // namespace mozilla
 
 #endif /* MOZILLA_AUDIOCHANNELFORMAT_H_ */

--- a/dom/media/AudioCompactor.cpp
+++ b/dom/media/AudioCompactor.cpp
@@ -1,62 +1,54 @@
 /* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
 /* vim: set ts=8 sts=2 et sw=2 tw=80: */
 /* This Source Code Form is subject to the terms of the Mozilla Public
  * License, v. 2.0. If a copy of the MPL was not distributed with this
  * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
 #include "AudioCompactor.h"
 #if defined(MOZ_MEMORY)
-# include "mozmemory.h"
+#include "mozmemory.h"
 #endif
 
 namespace mozilla {
 
-static size_t
-MallocGoodSize(size_t aSize)
-{
-# if defined(MOZ_MEMORY)
+static size_t MallocGoodSize(size_t aSize) {
+#if defined(MOZ_MEMORY)
   return malloc_good_size(aSize);
-# else
+#else
   return aSize;
-# endif
+#endif
 }
 
-static size_t
-TooMuchSlop(size_t aSize, size_t aAllocSize, size_t aMaxSlop)
-{
+static size_t TooMuchSlop(size_t aSize, size_t aAllocSize, size_t aMaxSlop) {
   // If the allocated size is less then our target size, then we
   // are chunking.  This means it will be completely filled with
   // zero slop.
   size_t slop = (aAllocSize > aSize) ? (aAllocSize - aSize) : 0;
   return slop > aMaxSlop;
 }
 
-uint32_t
-AudioCompactor::GetChunkSamples(uint32_t aFrames, uint32_t aChannels,
-                                size_t aMaxSlop)
-{
+uint32_t AudioCompactor::GetChunkSamples(uint32_t aFrames, uint32_t aChannels,
+                                         size_t aMaxSlop) {
   size_t size = AudioDataSize(aFrames, aChannels);
   size_t chunkSize = MallocGoodSize(size);
 
   // Reduce the chunk size until we meet our slop goal or the chunk
   // approaches an unreasonably small size.
   while (chunkSize > 64 && TooMuchSlop(size, chunkSize, aMaxSlop)) {
     chunkSize = MallocGoodSize(chunkSize / 2);
   }
 
   // Calculate the number of samples based on expected malloc size
   // in order to allow as many frames as possible to be packed.
   return chunkSize / sizeof(AudioDataValue);
 }
 
-uint32_t
-AudioCompactor::NativeCopy::operator()(AudioDataValue *aBuffer,
-                                       uint32_t aSamples)
-{
+uint32_t AudioCompactor::NativeCopy::operator()(AudioDataValue *aBuffer,
+                                                uint32_t aSamples) {
   NS_ASSERTION(aBuffer, "cannot copy to null buffer pointer");
   NS_ASSERTION(aSamples, "cannot copy zero values");
 
   size_t bufferBytes = aSamples * sizeof(AudioDataValue);
   size_t maxBytes = std::min(bufferBytes, mSourceBytes - mNextByte);
   uint32_t frames = maxBytes / BytesPerFrame(mChannels);
   size_t bytes = frames * BytesPerFrame(mChannels);
 
@@ -65,9 +57,9 @@ AudioCompactor::NativeCopy::operator()(A
   NS_ASSERTION(bytes <= bufferBytes, "tried to copy beyond destination buffer");
 
   memcpy(aBuffer, mSource + mNextByte, bytes);
 
   mNextByte += bytes;
   return frames;
 }
 
-} // namespace mozilla
+}  // namespace mozilla

--- a/dom/media/AudioCompactor.h
+++ b/dom/media/AudioCompactor.h
@@ -7,22 +7,19 @@
 #define AudioCompactor_h
 
 #include "MediaQueue.h"
 #include "MediaData.h"
 #include "VideoUtils.h"
 
 namespace mozilla {
 
-class AudioCompactor
-{
-public:
-  explicit AudioCompactor(MediaQueue<AudioData>& aQueue)
-    : mQueue(aQueue)
-  {
+class AudioCompactor {
+ public:
+  explicit AudioCompactor(MediaQueue<AudioData>& aQueue) : mQueue(aQueue) {
     // Determine padding size used by AlignedBuffer.
     size_t paddedSize = AlignedAudioBuffer::AlignmentPaddingSize();
     mSamplesPadding = paddedSize / sizeof(AudioDataValue);
     if (mSamplesPadding * sizeof(AudioDataValue) < paddedSize) {
       // Round up.
       mSamplesPadding++;
     }
   }
@@ -34,20 +31,19 @@ public:
   //
   //   uint32_t operator()(AudioDataValue *aBuffer, uint32_t aSamples);
   //
   // The functor must copy as many complete frames as possible to the provided
   // buffer given its length (in AudioDataValue elements).  The number of frames
   // copied must be returned.  This copy functor must support being called
   // multiple times in order to copy the audio data fully.  The copy functor
   // must copy full frames as partial frames will be ignored.
-  template<typename CopyFunc>
+  template <typename CopyFunc>
   bool Push(int64_t aOffset, int64_t aTime, int32_t aSampleRate,
-            uint32_t aFrames, uint32_t aChannels, CopyFunc aCopyFunc)
-  {
+            uint32_t aFrames, uint32_t aChannels, CopyFunc aCopyFunc) {
     auto time = media::TimeUnit::FromMicroseconds(aTime);
 
     // If we are losing more than a reasonable amount to padding, try to chunk
     // the data.
     size_t maxSlop = AudioDataSize(aFrames, aChannels) / MAX_SLOP_DIVISOR;
 
     while (aFrames > 0) {
       uint32_t samples = GetChunkSamples(aFrames, aChannels, maxSlop);
@@ -65,76 +61,66 @@ public:
       NS_ASSERTION(framesCopied <= aFrames, "functor copied too many frames");
       buffer.SetLength(size_t(framesCopied) * aChannels);
 
       auto duration = FramesToTimeUnit(framesCopied, aSampleRate);
       if (!duration.IsValid()) {
         return false;
       }
 
-      mQueue.Push(new AudioData(aOffset,
-                                time,
-                                duration,
-                                framesCopied,
-                                std::move(buffer),
-                                aChannels,
-                                aSampleRate));
+      mQueue.Push(new AudioData(aOffset, time, duration, framesCopied,
+                                std::move(buffer), aChannels, aSampleRate));
 
       // Remove the frames we just pushed into the queue and loop if there is
       // more to be done.
       time += duration;
       aFrames -= framesCopied;
 
       // NOTE: No need to update aOffset as its only an approximation anyway.
     }
 
     return true;
   }
 
   // Copy functor suitable for copying audio samples already in the
   // AudioDataValue format/layout expected by AudioStream on this platform.
-  class NativeCopy
-  {
-  public:
-    NativeCopy(const uint8_t* aSource, size_t aSourceBytes,
-               uint32_t aChannels)
-      : mSource(aSource)
-      , mSourceBytes(aSourceBytes)
-      , mChannels(aChannels)
-      , mNextByte(0)
-    { }
+  class NativeCopy {
+   public:
+    NativeCopy(const uint8_t* aSource, size_t aSourceBytes, uint32_t aChannels)
+        : mSource(aSource),
+          mSourceBytes(aSourceBytes),
+          mChannels(aChannels),
+          mNextByte(0) {}
 
-    uint32_t operator()(AudioDataValue *aBuffer, uint32_t aSamples);
+    uint32_t operator()(AudioDataValue* aBuffer, uint32_t aSamples);
 
-  private:
+   private:
     const uint8_t* const mSource;
     const size_t mSourceBytes;
     const uint32_t mChannels;
     size_t mNextByte;
   };
 
   // Allow 12.5% slop before chunking kicks in.  Public so that the gtest can
   // access it.
   static const size_t MAX_SLOP_DIVISOR = 8;
 
-private:
+ private:
   // Compute the number of AudioDataValue samples that will be fit the most
   // frames while keeping heap allocation slop less than the given threshold.
-  static uint32_t
-  GetChunkSamples(uint32_t aFrames, uint32_t aChannels, size_t aMaxSlop);
+  static uint32_t GetChunkSamples(uint32_t aFrames, uint32_t aChannels,
+                                  size_t aMaxSlop);
 
-  static size_t BytesPerFrame(uint32_t aChannels)
-  {
+  static size_t BytesPerFrame(uint32_t aChannels) {
     return sizeof(AudioDataValue) * aChannels;
   }
 
-  static size_t AudioDataSize(uint32_t aFrames, uint32_t aChannels)
-  {
+  static size_t AudioDataSize(uint32_t aFrames, uint32_t aChannels) {
     return aFrames * BytesPerFrame(aChannels);
   }
 
-  MediaQueue<AudioData> &mQueue;
+  MediaQueue<AudioData>& mQueue;
   size_t mSamplesPadding;
 };
 
-} // namespace mozilla
+}  // namespace mozilla
 
-#endif // AudioCompactor_h
+#endif  // AudioCompactor_h

--- a/dom/media/AudioConfig.cpp
+++ b/dom/media/AudioConfig.cpp
@@ -31,30 +31,26 @@ typedef AudioConfig::ChannelLayout Chann
  2F2            L   R   LS   RS
  2F2-LFE        L   R   LFE  LS   RS
  3F2            L   R   C    LS   RS
  3F2-LFE        L   R   C    LFE  LS   RS
  3F3R-LFE       L   R   C    LFE  BC   LS   RS
  3F4-LFE        L   R   C    LFE  Rls  Rrs  LS   RS
 */
 
-void
-AudioConfig::ChannelLayout::UpdateChannelMap()
-{
+void AudioConfig::ChannelLayout::UpdateChannelMap() {
   mValid = mChannels.Length() <= MAX_CHANNELS;
   mChannelMap = UNKNOWN_MAP;
   if (mValid) {
     mChannelMap = Map();
     mValid = mChannelMap > 0;
   }
 }
 
-auto
-AudioConfig::ChannelLayout::Map() const -> ChannelMap
-{
+auto AudioConfig::ChannelLayout::Map() const -> ChannelMap {
   if (mChannelMap != UNKNOWN_MAP) {
     return mChannelMap;
   }
   if (mChannels.Length() > MAX_CHANNELS) {
     return UNKNOWN_MAP;
   }
   ChannelMap map = UNKNOWN_MAP;
   for (size_t i = 0; i < mChannels.Length(); i++) {
@@ -67,164 +63,148 @@ AudioConfig::ChannelLayout::Map() const 
       return UNKNOWN_MAP;
     }
     map |= mask;
   }
   return map;
 }
 
 const AudioConfig::Channel*
-AudioConfig::ChannelLayout::DefaultLayoutForChannels(uint32_t aChannels) const
-{
+AudioConfig::ChannelLayout::DefaultLayoutForChannels(uint32_t aChannels) const {
   switch (aChannels) {
-    case 1: // MONO
+    case 1:  // MONO
     {
-      static const Channel config[] = { CHANNEL_FRONT_CENTER };
+      static const Channel config[] = {CHANNEL_FRONT_CENTER};
       return config;
     }
-    case 2: // STEREO
+    case 2:  // STEREO
     {
-      static const Channel config[] = { CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT };
+      static const Channel config[] = {CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT};
       return config;
     }
-    case 3: // 3F
+    case 3:  // 3F
     {
-      static const Channel config[] = { CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT, CHANNEL_FRONT_CENTER };
+      static const Channel config[] = {CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT,
+                                       CHANNEL_FRONT_CENTER};
+      return config;
+    }
+    case 4:  // QUAD
+    {
+      static const Channel config[] = {CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT,
+                                       CHANNEL_BACK_LEFT, CHANNEL_BACK_RIGHT};
       return config;
     }
-    case 4: // QUAD
+    case 5:  // 3F2
     {
-      static const Channel config[] = { CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT, CHANNEL_BACK_LEFT, CHANNEL_BACK_RIGHT };
+      static const Channel config[] = {CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT,
+                                       CHANNEL_FRONT_CENTER, CHANNEL_SIDE_LEFT,
+                                       CHANNEL_SIDE_RIGHT};
       return config;
     }
-    case 5: // 3F2
+    case 6:  // 3F2-LFE
     {
-      static const Channel config[] = { CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT, CHANNEL_FRONT_CENTER, CHANNEL_SIDE_LEFT, CHANNEL_SIDE_RIGHT };
+      static const Channel config[] = {
+          CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT, CHANNEL_FRONT_CENTER,
+          CHANNEL_LFE,        CHANNEL_SIDE_LEFT,   CHANNEL_SIDE_RIGHT};
       return config;
     }
-    case 6: // 3F2-LFE
+    case 7:  // 3F3R-LFE
     {
-      static const Channel config[] = { CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT, CHANNEL_FRONT_CENTER, CHANNEL_LFE, CHANNEL_SIDE_LEFT, CHANNEL_SIDE_RIGHT };
+      static const Channel config[] = {
+          CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT, CHANNEL_FRONT_CENTER,
+          CHANNEL_LFE,        CHANNEL_BACK_CENTER, CHANNEL_SIDE_LEFT,
+          CHANNEL_SIDE_RIGHT};
       return config;
     }
-    case 7: // 3F3R-LFE
+    case 8:  // 3F4-LFE
     {
-      static const Channel config[] = { CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT, CHANNEL_FRONT_CENTER, CHANNEL_LFE, CHANNEL_BACK_CENTER, CHANNEL_SIDE_LEFT, CHANNEL_SIDE_RIGHT };
-      return config;
-    }
-    case 8: // 3F4-LFE
-    {
-      static const Channel config[] = { CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT, CHANNEL_FRONT_CENTER, CHANNEL_LFE, CHANNEL_BACK_LEFT, CHANNEL_BACK_RIGHT, CHANNEL_SIDE_LEFT, CHANNEL_SIDE_RIGHT };
+      static const Channel config[] = {
+          CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT, CHANNEL_FRONT_CENTER,
+          CHANNEL_LFE,        CHANNEL_BACK_LEFT,   CHANNEL_BACK_RIGHT,
+          CHANNEL_SIDE_LEFT,  CHANNEL_SIDE_RIGHT};
       return config;
     }
     default:
       return nullptr;
   }
 }
 
 /* static */ AudioConfig::ChannelLayout
-AudioConfig::ChannelLayout::SMPTEDefault(
-  const ChannelLayout& aChannelLayout)
-{
+AudioConfig::ChannelLayout::SMPTEDefault(const ChannelLayout& aChannelLayout) {
   if (!aChannelLayout.IsValid()) {
     return aChannelLayout;
   }
   return SMPTEDefault(aChannelLayout.Map());
 }
 
-/* static */ ChannelLayout
-AudioConfig::ChannelLayout::SMPTEDefault(ChannelMap aMap)
-{
+/* static */ ChannelLayout AudioConfig::ChannelLayout::SMPTEDefault(
+    ChannelMap aMap) {
   // First handle the most common cases.
   switch (aMap) {
     case LMONO_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_CENTER };
+      return ChannelLayout{CHANNEL_FRONT_CENTER};
     case LSTEREO_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT };
+      return ChannelLayout{CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT};
     case L3F_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_LEFT,
-                            CHANNEL_FRONT_RIGHT,
-                            CHANNEL_FRONT_CENTER };
+      return ChannelLayout{CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT,
+                           CHANNEL_FRONT_CENTER};
     case L3F_LFE_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_LEFT,
-                            CHANNEL_FRONT_RIGHT,
-                            CHANNEL_FRONT_CENTER,
-                            CHANNEL_LFE };
+      return ChannelLayout{CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT,
+                           CHANNEL_FRONT_CENTER, CHANNEL_LFE};
     case L2F1_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_LEFT,
-                            CHANNEL_FRONT_RIGHT,
-                            CHANNEL_BACK_CENTER };
+      return ChannelLayout{CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT,
+                           CHANNEL_BACK_CENTER};
     case L2F1_LFE_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_LEFT,
-                            CHANNEL_FRONT_RIGHT,
-                            CHANNEL_LFE,
-                            CHANNEL_BACK_CENTER };
+      return ChannelLayout{CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT, CHANNEL_LFE,
+                           CHANNEL_BACK_CENTER};
     case L3F1_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_LEFT,
-                            CHANNEL_FRONT_RIGHT,
-                            CHANNEL_FRONT_CENTER,
-                            CHANNEL_BACK_CENTER };
+      return ChannelLayout{CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT,
+                           CHANNEL_FRONT_CENTER, CHANNEL_BACK_CENTER};
     case L3F1_LFE_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_LEFT,
-                            CHANNEL_FRONT_RIGHT,
-                            CHANNEL_FRONT_CENTER,
-                            CHANNEL_LFE,
-                            CHANNEL_BACK_CENTER };
+      return ChannelLayout{CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT,
+                           CHANNEL_FRONT_CENTER, CHANNEL_LFE,
+                           CHANNEL_BACK_CENTER};
     case L2F2_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_LEFT,
-                            CHANNEL_FRONT_RIGHT,
-                            CHANNEL_SIDE_LEFT,
-                            CHANNEL_SIDE_RIGHT };
+      return ChannelLayout{CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT,
+                           CHANNEL_SIDE_LEFT, CHANNEL_SIDE_RIGHT};
     case L2F2_LFE_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_LEFT,
-                            CHANNEL_FRONT_RIGHT,
-                            CHANNEL_LFE,
-                            CHANNEL_SIDE_LEFT,
-                            CHANNEL_SIDE_RIGHT };
+      return ChannelLayout{CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT, CHANNEL_LFE,
+                           CHANNEL_SIDE_LEFT, CHANNEL_SIDE_RIGHT};
     case LQUAD_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_LEFT,
-                            CHANNEL_FRONT_RIGHT,
-                            CHANNEL_BACK_LEFT,
-                            CHANNEL_BACK_RIGHT };
+      return ChannelLayout{CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT,
+                           CHANNEL_BACK_LEFT, CHANNEL_BACK_RIGHT};
     case LQUAD_LFE_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_LEFT,
-                            CHANNEL_FRONT_RIGHT,
-                            CHANNEL_LFE,
-                            CHANNEL_BACK_LEFT,
-                            CHANNEL_BACK_RIGHT };
+      return ChannelLayout{CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT, CHANNEL_LFE,
+                           CHANNEL_BACK_LEFT, CHANNEL_BACK_RIGHT};
     case L3F2_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_LEFT,
-                            CHANNEL_FRONT_RIGHT,
-                            CHANNEL_FRONT_CENTER,
-                            CHANNEL_SIDE_LEFT,
-                            CHANNEL_SIDE_RIGHT };
+      return ChannelLayout{CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT,
+                           CHANNEL_FRONT_CENTER, CHANNEL_SIDE_LEFT,
+                           CHANNEL_SIDE_RIGHT};
     case L3F2_LFE_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_LEFT,   CHANNEL_FRONT_RIGHT,
-                            CHANNEL_FRONT_CENTER, CHANNEL_LFE,
-                            CHANNEL_SIDE_LEFT,    CHANNEL_SIDE_RIGHT };
+      return ChannelLayout{CHANNEL_FRONT_LEFT,   CHANNEL_FRONT_RIGHT,
+                           CHANNEL_FRONT_CENTER, CHANNEL_LFE,
+                           CHANNEL_SIDE_LEFT,    CHANNEL_SIDE_RIGHT};
     case L3F2_BACK_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_LEFT,
-                            CHANNEL_FRONT_RIGHT,
-                            CHANNEL_FRONT_CENTER,
-                            CHANNEL_BACK_LEFT,
-                            CHANNEL_BACK_RIGHT };
+      return ChannelLayout{CHANNEL_FRONT_LEFT, CHANNEL_FRONT_RIGHT,
+                           CHANNEL_FRONT_CENTER, CHANNEL_BACK_LEFT,
+                           CHANNEL_BACK_RIGHT};
     case L3F2_BACK_LFE_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_LEFT,   CHANNEL_FRONT_RIGHT,
-                            CHANNEL_FRONT_CENTER, CHANNEL_LFE,
-                            CHANNEL_BACK_LEFT,    CHANNEL_BACK_RIGHT };
+      return ChannelLayout{CHANNEL_FRONT_LEFT,   CHANNEL_FRONT_RIGHT,
+                           CHANNEL_FRONT_CENTER, CHANNEL_LFE,
+                           CHANNEL_BACK_LEFT,    CHANNEL_BACK_RIGHT};
     case L3F3R_LFE_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_LEFT,   CHANNEL_FRONT_RIGHT,
-                            CHANNEL_FRONT_CENTER, CHANNEL_LFE,
-                            CHANNEL_BACK_CENTER,  CHANNEL_SIDE_LEFT,
-                            CHANNEL_SIDE_RIGHT };
+      return ChannelLayout{CHANNEL_FRONT_LEFT,   CHANNEL_FRONT_RIGHT,
+                           CHANNEL_FRONT_CENTER, CHANNEL_LFE,
+                           CHANNEL_BACK_CENTER,  CHANNEL_SIDE_LEFT,
+                           CHANNEL_SIDE_RIGHT};
     case L3F4_LFE_MAP:
-      return ChannelLayout{ CHANNEL_FRONT_LEFT,   CHANNEL_FRONT_RIGHT,
-                            CHANNEL_FRONT_CENTER, CHANNEL_LFE,
-                            CHANNEL_BACK_LEFT,    CHANNEL_BACK_RIGHT,
-                            CHANNEL_SIDE_LEFT,    CHANNEL_SIDE_RIGHT };
+      return ChannelLayout{CHANNEL_FRONT_LEFT,   CHANNEL_FRONT_RIGHT,
+                           CHANNEL_FRONT_CENTER, CHANNEL_LFE,
+                           CHANNEL_BACK_LEFT,    CHANNEL_BACK_RIGHT,
+                           CHANNEL_SIDE_LEFT,    CHANNEL_SIDE_RIGHT};
     default:
       break;
   }
 
   static_assert(MAX_CHANNELS <= sizeof(ChannelMap) * 8,
                 "Must be able to fit channels on bit mask");
   AutoTArray<Channel, MAX_CHANNELS> layout;
   uint32_t channels = 0;
@@ -239,20 +219,18 @@ AudioConfig::ChannelLayout::SMPTEDefault
       layout.AppendElement(static_cast<Channel>(i));
     }
     aMap >>= 1;
     i++;
   }
   return ChannelLayout(channels, layout.Elements());
 }
 
-bool
-AudioConfig::ChannelLayout::MappingTable(const ChannelLayout& aOther,
-                                         nsTArray<uint8_t>* aMap) const
-{
+bool AudioConfig::ChannelLayout::MappingTable(const ChannelLayout& aOther,
+                                              nsTArray<uint8_t>* aMap) const {
   if (!IsValid() || !aOther.IsValid() || Map() != aOther.Map()) {
     if (aMap) {
       aMap->SetLength(0);
     }
     return false;
   }
   if (!aMap) {
     return true;
@@ -268,88 +246,98 @@ AudioConfig::ChannelLayout::MappingTable
   }
   return true;
 }
 
 /**
  * AudioConfig::ChannelConfig
  */
 
-/* static */ const char*
-AudioConfig::FormatToString(AudioConfig::SampleFormat aFormat)
-{
+/* static */ const char* AudioConfig::FormatToString(
+    AudioConfig::SampleFormat aFormat) {
   switch (aFormat) {
-    case FORMAT_U8:     return "unsigned 8 bit";
-    case FORMAT_S16:    return "signed 16 bit";
-    case FORMAT_S24:    return "signed 24 bit MSB";
-    case FORMAT_S24LSB: return "signed 24 bit LSB";
-    case FORMAT_S32:    return "signed 32 bit";
-    case FORMAT_FLT:    return "32 bit floating point";
-    case FORMAT_NONE:   return "none";
-    default:            return "unknown";
+    case FORMAT_U8:
+      return "unsigned 8 bit";
+    case FORMAT_S16:
+      return "signed 16 bit";
+    case FORMAT_S24:
+      return "signed 24 bit MSB";
+    case FORMAT_S24LSB:
+      return "signed 24 bit LSB";
+    case FORMAT_S32:
+      return "signed 32 bit";
+    case FORMAT_FLT:
+      return "32 bit floating point";
+    case FORMAT_NONE:
+      return "none";
+    default:
+      return "unknown";
   }
 }
-/* static */ uint32_t
-AudioConfig::SampleSize(AudioConfig::SampleFormat aFormat)
-{
+/* static */ uint32_t AudioConfig::SampleSize(
+    AudioConfig::SampleFormat aFormat) {
   switch (aFormat) {
-    case FORMAT_U8:     return 1;
-    case FORMAT_S16:    return 2;
-    case FORMAT_S24:    MOZ_FALLTHROUGH;
-    case FORMAT_S24LSB: MOZ_FALLTHROUGH;
-    case FORMAT_S32:    MOZ_FALLTHROUGH;
-    case FORMAT_FLT:    return 4;
+    case FORMAT_U8:
+      return 1;
+    case FORMAT_S16:
+      return 2;
+    case FORMAT_S24:
+      MOZ_FALLTHROUGH;
+    case FORMAT_S24LSB:
+      MOZ_FALLTHROUGH;
+    case FORMAT_S32:
+      MOZ_FALLTHROUGH;
+    case FORMAT_FLT:
+      return 4;
     case FORMAT_NONE:
-    default:            return 0;
+    default:
+      return 0;
   }
 }
 
-/* static */ uint32_t
-AudioConfig::FormatToBits(AudioConfig::SampleFormat aFormat)
-{
+/* static */ uint32_t AudioConfig::FormatToBits(
+    AudioConfig::SampleFormat aFormat) {
   switch (aFormat) {
-    case FORMAT_U8:     return 8;
-    case FORMAT_S16:    return 16;
-    case FORMAT_S24LSB: MOZ_FALLTHROUGH;
-    case FORMAT_S24:    return 24;
-    case FORMAT_S32:    MOZ_FALLTHROUGH;
-    case FORMAT_FLT:    return 32;
-    case FORMAT_NONE:   MOZ_FALLTHROUGH;
-    default:            return 0;
+    case FORMAT_U8:
+      return 8;
+    case FORMAT_S16:
+      return 16;
+    case FORMAT_S24LSB:
+      MOZ_FALLTHROUGH;
+    case FORMAT_S24:
+      return 24;
+    case FORMAT_S32:
+      MOZ_FALLTHROUGH;
+    case FORMAT_FLT:
+      return 32;
+    case FORMAT_NONE:
+      MOZ_FALLTHROUGH;
+    default:
+      return 0;
   }
 }
 
 AudioConfig::AudioConfig(const ChannelLayout& aChannelLayout, uint32_t aRate,
                          AudioConfig::SampleFormat aFormat, bool aInterleaved)
-  : mChannelLayout(aChannelLayout)
-  , mChannels(aChannelLayout.Count())
-  , mRate(aRate)
-  , mFormat(aFormat)
-  , mInterleaved(aInterleaved)
-{
-}
+    : mChannelLayout(aChannelLayout),
+      mChannels(aChannelLayout.Count()),
+      mRate(aRate),
+      mFormat(aFormat),
+      mInterleaved(aInterleaved) {}
 
 AudioConfig::AudioConfig(const ChannelLayout& aChannelLayout,
-                         uint32_t aChannels,
-                         uint32_t aRate,
-                         AudioConfig::SampleFormat aFormat,
-                         bool aInterleaved)
-  : mChannelLayout(aChannelLayout)
-  , mChannels(aChannels)
-  , mRate(aRate)
-  , mFormat(aFormat)
-  , mInterleaved(aInterleaved)
-{
-}
+                         uint32_t aChannels, uint32_t aRate,
+                         AudioConfig::SampleFormat aFormat, bool aInterleaved)
+    : mChannelLayout(aChannelLayout),
+      mChannels(aChannels),
+      mRate(aRate),
+      mFormat(aFormat),
+      mInterleaved(aInterleaved) {}
 
-AudioConfig::AudioConfig(uint32_t aChannels,
-                         uint32_t aRate,
-                         AudioConfig::SampleFormat aFormat,
-                         bool aInterleaved)
-  : mChannelLayout(aChannels)
-  , mChannels(aChannels)
-  , mRate(aRate)
-  , mFormat(aFormat)
-  , mInterleaved(aInterleaved)
-{
-}
+AudioConfig::AudioConfig(uint32_t aChannels, uint32_t aRate,
+                         AudioConfig::SampleFormat aFormat, bool aInterleaved)
+    : mChannelLayout(aChannels),
+      mChannels(aChannels),
+      mRate(aRate),
+      mFormat(aFormat),
+      mInterleaved(aInterleaved) {}
 
-} // namespace mozilla

author	Sylvestre Ledru <sledru@mozilla.com>
	Mon, 19 Nov 2018 13:25:37 +0000
changeset 446960	0ceae9db9ec0be18daa1a279511ad305723185d4
parent 446959	aed811658fae595beec6344b597ab8bca2ac9c83
child 446992	62c4741625fed62804f105d8d979f4b5b3d0f191
child 446993	c8dd8f4166c9287abf6964e55f3b86754d6cf806
push id	35063
push user	dvarga@mozilla.com
push date	Mon, 19 Nov 2018 16:59:56 +0000
treeherder	mozilla-central@0ceae9db9ec0 [default view] [failures only]
perfherder	[talos] [build metrics] [platform microbench] (compared to previous push)
reviewers	jya
bugs	1204606
milestone	65.0a1
first release with	nightly linux32 bd4cebdbed4b / 65.0a1 / 20181119220031 / files nightly linux64 bd4cebdbed4b / 65.0a1 / 20181119220031 / files nightly mac bd4cebdbed4b / 65.0a1 / 20181119220031 / files nightly win32 bd4cebdbed4b / 65.0a1 / 20181119220031 / files nightly win64 bd4cebdbed4b / 65.0a1 / 20181119220031 / files
last release without	nightly linux32 e44bb5b4bc79 / 65.0a1 / 20181119100448 / files nightly linux64 e44bb5b4bc79 / 65.0a1 / 20181119100448 / files nightly mac e44bb5b4bc79 / 65.0a1 / 20181119100448 / files nightly win32 e44bb5b4bc79 / 65.0a1 / 20181119100448 / files nightly win64 e44bb5b4bc79 / 65.0a1 / 20181119100448 / files