Merge branch 'master' into Pale_Moon-release

# Conflicts:
#	application/palemoon/config/version.txt
#	js/src/jit/x86-shared/AssemblerBuffer-x86-shared.h
#	toolkit/components/search/orginal/nsSearchService.js

8 files changed, 44 insertions, 145 deletions

diff --git a/media/ffvpx/README_MCP b/media/ffvpx/README_MCP
index 4555e08da..5ed4d8d94 100644
--- a/media/ffvpx/README_MCP
+++ b/media/ffvpx/README_MCP
@@ -7,10 +7,6 @@ Currently, we only use the vp8 and vp9 portion of the library, and only on x86
 based platforms. If this changes, configuration files will most likely
 need to be updated.
 
-AVX2 must be disabled on Linux 32 bits due to the use of yasm 1.1 on the build bots.
-Once yasm is upgraded to 1.2 or later, AVX2 code could be re-enabled.
-Add --disable-avx2 to configure on those platforms.
-
 configuration files were generated as follow using the configure script:
 ./configure --disable-everything --disable-protocols --disable-demuxers --disable-muxers --disable-filters --disable-programs --disable-doc --disable-parsers --enable-parser=vp8 --enable-parser=vp9 --enable-decoder=vp8 --enable-decoder=vp9 --disable-static --enable-shared --disable-debug --disable-sdl2 --disable-libxcb --disable-securetransport --disable-iconv --disable-swresample --disable-swscale --disable-avdevice --disable-avfilter --disable-avformat --disable-d3d11va --disable-dxva2 --disable-vaapi --disable-vda --disable-vdpau --disable-videotoolbox --enable-decoder=flac --enable-asm --enable-x86asm
 
@@ -18,13 +14,12 @@ config*:
 replace: /HAVE_(MALLOC_H|ARC4RANDOM|LOCALTIME_R|MEMALIGN|POSIX_MEMALIGN)/d
 
 config_darwin32.h:
-add to configure command: --disable-asm --disable-yasm --cc='clang -m32'
+add to configure command: --disable-asm --disable-x86asm --cc='clang -m32'
 
 config_unix32.h:
 add to configure command: --disable-asm --disable-x86asm --cc='clang -m32'
 replace: s/HAVE_SYSCTL 1/HAVE_SYSCTL 0/ and s/HAVE_MEMALIGN 1/HAVE_MEMALIGN 0/ and s/HAVE_POSIX_MEMALIGN 1/HAVE_POSIX_MEMALIGN 0/
 
-
 config_unix32.h/config_unix64.h/config_unix64.asm:
 replace: s/HAVE_SYSCTL 1/HAVE_SYSCTL 0
 
diff --git a/media/libvpx/bug1480092.patch b/media/libvpx/bug1480092.patch
new file mode 100644
index 000000000..ae774bb20
--- /dev/null
+++ b/media/libvpx/bug1480092.patch
@@ -0,0 +1,36 @@
+From f00fe25d7eb13ceafbea6a6987d45fdef64cffb3 Mon Sep 17 00:00:00 2001
+From: Pale Moon <git-repo@palemoon.org>
+Date: Tue, 11 Sep 2018 08:58:16 +0200
+Subject: [PATCH] Cherry-pick libvpx upstream
+ 52add5896661d186dec284ed646a4b33b607d2c7.
+
+---
+ media/libvpx/vp8/common/postproc.c | 2 +-
+ 1 file changed, 1 insertion(+), 1 deletion(-)
+
+diff --git a/media/libvpx/vp8/common/postproc.c b/media/libvpx/vp8/common/postproc.c
+index a4e6ae170..3b05bc63e 100644
+--- a/media/libvpx/vp8/common/postproc.c
++++ b/media/libvpx/vp8/common/postproc.c
+@@ -325,17 +325,17 @@ void vp8_deblock(VP8_COMMON                 *cm,
+                  YV12_BUFFER_CONFIG         *post,
+                  int                         q,
+                  int                         low_var_thresh,
+                  int                         flag)
+ {
+     double level = 6.0e-05 * q * q * q - .0067 * q * q + .306 * q + .0065;
+     int ppl = (int)(level + .5);
+ 
+-    const MODE_INFO *mode_info_context = cm->show_frame_mi;
++    const MODE_INFO *mode_info_context = cm->mi;
+     int mbr, mbc;
+ 
+     /* The pixel thresholds are adjusted according to if or not the macroblock
+      * is a skipped block.  */
+     unsigned char *ylimits = cm->pp_limits_buffer;
+     unsigned char *uvlimits = cm->pp_limits_buffer + 16 * cm->mb_cols;
+     (void) low_var_thresh;
+     (void) flag;
+-- 
+2.16.1.windows.4
+
diff --git a/media/libvpx/update.py b/media/libvpx/update.py
index 85eed5872..1e9d9b478 100755
--- a/media/libvpx/update.py
+++ b/media/libvpx/update.py
@@ -608,6 +608,8 @@ def apply_patches():
     os.system("patch -p3 < input_frame_validation.patch")
     # Bug 1315288 - Check input frame resolution for vp9
     os.system("patch -p3 < input_frame_validation_vp9.patch")
+    # Cherrypick fix from upstream
+    os.system("patch -p3 < bug1480092.patch")
 
 def update_readme(commit):
     with open('README_MOZILLA') as f:
diff --git a/media/libvpx/vp8/common/postproc.c b/media/libvpx/vp8/common/postproc.c
index a4e6ae170..3b05bc63e 100644
--- a/media/libvpx/vp8/common/postproc.c
+++ b/media/libvpx/vp8/common/postproc.c
@@ -330,7 +330,7 @@ void vp8_deblock(VP8_COMMON                 *cm,
     double level = 6.0e-05 * q * q * q - .0067 * q * q + .306 * q + .0065;
     int ppl = (int)(level + .5);
 
-    const MODE_INFO *mode_info_context = cm->show_frame_mi;
+    const MODE_INFO *mode_info_context = cm->mi;
     int mbr, mbc;
 
     /* The pixel thresholds are adjusted according to if or not the macroblock
diff --git a/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp b/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
index 2c57431e7..e36b8b6cf 100755
--- a/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
+++ b/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
@@ -706,15 +706,6 @@ WebrtcAudioConduit::GetAudioFrame(int16_t speechData[],
     if (GetAVStats(&jitter_buffer_delay_ms,
                    &playout_buffer_delay_ms,
                    &avsync_offset_ms)) {
-#if !defined(MOZILLA_EXTERNAL_LINKAGE)
-      if (avsync_offset_ms < 0) {
-        Telemetry::Accumulate(Telemetry::WEBRTC_AVSYNC_WHEN_VIDEO_LAGS_AUDIO_MS,
-                              -avsync_offset_ms);
-      } else {
-        Telemetry::Accumulate(Telemetry::WEBRTC_AVSYNC_WHEN_AUDIO_LAGS_VIDEO_MS,
-                              avsync_offset_ms);
-      }
-#endif
       CSFLogError(logTag,
                   "A/V sync: sync delta: %dms, audio jitter delay %dms, playout delay %dms",
                   avsync_offset_ms, jitter_buffer_delay_ms, playout_buffer_delay_ms);
diff --git a/media/webrtc/signaling/src/media-conduit/CodecStatistics.cpp b/media/webrtc/signaling/src/media-conduit/CodecStatistics.cpp
index eb03c0bf8..da40a59ea 100644
--- a/media/webrtc/signaling/src/media-conduit/CodecStatistics.cpp
+++ b/media/webrtc/signaling/src/media-conduit/CodecStatistics.cpp
@@ -124,8 +124,6 @@ void VideoCodecStatistics::ReceiveStateChange(const int aChannel,
           TimeDuration timeDelta = TimeStamp::Now() - mReceiveFailureTime;
           CSFLogError(logTag, "Video error duration: %u ms",
                       static_cast<uint32_t>(timeDelta.ToMilliseconds()));
-          Telemetry::Accumulate(Telemetry::WEBRTC_VIDEO_ERROR_RECOVERY_MS,
-                                static_cast<uint32_t>(timeDelta.ToMilliseconds()));
 
           mRecoveredLosses++; // to calculate losses per minute
           mTotalLossTime += timeDelta;  // To calculate % time in recovery
@@ -147,16 +145,10 @@ void VideoCodecStatistics::EndOfCallStats()
     if (callDelta.ToSeconds() != 0) {
       uint32_t recovered_per_min = mRecoveredBeforeLoss/(callDelta.ToSeconds()/60);
       CSFLogError(logTag, "Video recovery before error per min %u", recovered_per_min);
-      Telemetry::Accumulate(Telemetry::WEBRTC_VIDEO_RECOVERY_BEFORE_ERROR_PER_MIN,
-                            recovered_per_min);
       uint32_t err_per_min = mRecoveredLosses/(callDelta.ToSeconds()/60);
       CSFLogError(logTag, "Video recovery after error per min %u", err_per_min);
-      Telemetry::Accumulate(Telemetry::WEBRTC_VIDEO_RECOVERY_AFTER_ERROR_PER_MIN,
-                            err_per_min);
       float percent = (mTotalLossTime.ToSeconds()*100)/callDelta.ToSeconds();
       CSFLogError(logTag, "Video error time percentage %f%%", percent);
-      Telemetry::Accumulate(Telemetry::WEBRTC_VIDEO_DECODE_ERROR_TIME_PERMILLE,
-                            static_cast<uint32_t>(percent*10));
     }
   }
 #endif
diff --git a/media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp b/media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp
index 33422ed7a..43d10ca86 100644
--- a/media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp
+++ b/media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp
@@ -2245,22 +2245,6 @@ PeerConnectionImpl::AddIceCandidate(const char* aCandidate, const char* aMid, un
 
   CSFLogDebug(logTag, "AddIceCandidate: %s", aCandidate);
 
-#if !defined(MOZILLA_EXTERNAL_LINKAGE)
-  // When remote candidates are added before our ICE ctx is up and running
-  // (the transition to New is async through STS, so this is not impossible),
-  // we won't record them as trickle candidates. Is this what we want?
-  if(!mIceStartTime.IsNull()) {
-    TimeDuration timeDelta = TimeStamp::Now() - mIceStartTime;
-    if (mIceConnectionState == PCImplIceConnectionState::Failed) {
-      Telemetry::Accumulate(Telemetry::WEBRTC_ICE_LATE_TRICKLE_ARRIVAL_TIME,
-                            timeDelta.ToMilliseconds());
-    } else {
-      Telemetry::Accumulate(Telemetry::WEBRTC_ICE_ON_TIME_TRICKLE_ARRIVAL_TIME,
-                            timeDelta.ToMilliseconds());
-    }
-  }
-#endif
-
   nsresult res = mJsepSession->AddRemoteIceCandidate(aCandidate, aMid, aLevel);
 
   if (NS_SUCCEEDED(res)) {
@@ -3011,49 +2995,7 @@ PeerConnectionImpl::PluginCrash(uint32_t aPluginID,
 void
 PeerConnectionImpl::RecordEndOfCallTelemetry() const
 {
-  if (!mJsepSession) {
-    return;
-  }
-
-#if !defined(MOZILLA_EXTERNAL_LINKAGE)
-  // Bitmask used for WEBRTC/LOOP_CALL_TYPE telemetry reporting
-  static const uint32_t kAudioTypeMask = 1;
-  static const uint32_t kVideoTypeMask = 2;
-  static const uint32_t kDataChannelTypeMask = 4;
-
-  // Report end-of-call Telemetry
-  if (mJsepSession->GetNegotiations() > 0) {
-    Telemetry::Accumulate(Telemetry::WEBRTC_RENEGOTIATIONS,
-                          mJsepSession->GetNegotiations()-1);
-  }
-  Telemetry::Accumulate(Telemetry::WEBRTC_MAX_VIDEO_SEND_TRACK,
-                        mMaxSending[SdpMediaSection::MediaType::kVideo]);
-  Telemetry::Accumulate(Telemetry::WEBRTC_MAX_VIDEO_RECEIVE_TRACK,
-                        mMaxReceiving[SdpMediaSection::MediaType::kVideo]);
-  Telemetry::Accumulate(Telemetry::WEBRTC_MAX_AUDIO_SEND_TRACK,
-                        mMaxSending[SdpMediaSection::MediaType::kAudio]);
-  Telemetry::Accumulate(Telemetry::WEBRTC_MAX_AUDIO_RECEIVE_TRACK,
-                        mMaxReceiving[SdpMediaSection::MediaType::kAudio]);
-  // DataChannels appear in both Sending and Receiving
-  Telemetry::Accumulate(Telemetry::WEBRTC_DATACHANNEL_NEGOTIATED,
-                        mMaxSending[SdpMediaSection::MediaType::kApplication]);
-  // Enumerated/bitmask: 1 = Audio, 2 = Video, 4 = DataChannel
-  // A/V = 3, A/V/D = 7, etc
-  uint32_t type = 0;
-  if (mMaxSending[SdpMediaSection::MediaType::kAudio] ||
-      mMaxReceiving[SdpMediaSection::MediaType::kAudio]) {
-    type = kAudioTypeMask;
-  }
-  if (mMaxSending[SdpMediaSection::MediaType::kVideo] ||
-      mMaxReceiving[SdpMediaSection::MediaType::kVideo]) {
-    type |= kVideoTypeMask;
-  }
-  if (mMaxSending[SdpMediaSection::MediaType::kApplication]) {
-    type |= kDataChannelTypeMask;
-  }
-  Telemetry::Accumulate(Telemetry::WEBRTC_CALL_TYPE,
-                        type);
-#endif
+  /* STUB */
 }
 
 nsresult
@@ -3109,13 +3051,6 @@ PeerConnectionImpl::ShutdownMedia()
       pair.second->RemovePrincipalChangeObserver(this);
     }
   }
-
-  // End of call to be recorded in Telemetry
-  if (!mStartTime.IsNull()){
-    TimeDuration timeDelta = TimeStamp::Now() - mStartTime;
-    Telemetry::Accumulate(Telemetry::WEBRTC_CALL_DURATION,
-                          timeDelta.ToSeconds());
-  }
 #endif
 
   // Forget the reference so that we can transfer it to
@@ -3423,33 +3358,6 @@ void PeerConnectionImpl::IceConnectionStateChange(
     return;
   }
 
-#if !defined(MOZILLA_EXTERNAL_LINKAGE)
-  if (!isDone(mIceConnectionState) && isDone(domState)) {
-    // mIceStartTime can be null if going directly from New to Closed, in which
-    // case we don't count it as a success or a failure.
-    if (!mIceStartTime.IsNull()){
-      TimeDuration timeDelta = TimeStamp::Now() - mIceStartTime;
-      if (isSucceeded(domState)) {
-        Telemetry::Accumulate(Telemetry::WEBRTC_ICE_SUCCESS_TIME,
-                              timeDelta.ToMilliseconds());
-      } else if (isFailed(domState)) {
-        Telemetry::Accumulate(Telemetry::WEBRTC_ICE_FAILURE_TIME,
-                              timeDelta.ToMilliseconds());
-      }
-    }
-
-    if (isSucceeded(domState)) {
-      Telemetry::Accumulate(
-          Telemetry::WEBRTC_ICE_ADD_CANDIDATE_ERRORS_GIVEN_SUCCESS,
-          mAddCandidateErrorCount);
-    } else if (isFailed(domState)) {
-      Telemetry::Accumulate(
-          Telemetry::WEBRTC_ICE_ADD_CANDIDATE_ERRORS_GIVEN_FAILURE,
-          mAddCandidateErrorCount);
-    }
-  }
-#endif
-
   mIceConnectionState = domState;
 
   if (mIceConnectionState == PCImplIceConnectionState::Connected ||
@@ -3467,10 +3375,6 @@ void PeerConnectionImpl::IceConnectionStateChange(
       STAMP_TIMECARD(mTimeCard, "Ice state: new");
       break;
     case PCImplIceConnectionState::Checking:
-#if !defined(MOZILLA_EXTERNAL_LINKAGE)
-      // For telemetry
-      mIceStartTime = TimeStamp::Now();
-#endif
       STAMP_TIMECARD(mTimeCard, "Ice state: checking");
       break;
     case PCImplIceConnectionState::Connected:
@@ -4067,16 +3971,7 @@ PeerConnectionImpl::IceStreamReady(NrIceMediaStream *aStream)
 //Telemetry for when calls start
 void
 PeerConnectionImpl::startCallTelem() {
-  if (!mStartTime.IsNull()) {
-    return;
-  }
-
-  // Start time for calls
-  mStartTime = TimeStamp::Now();
-
-  // Increment session call counter
-  // If we want to track Loop calls independently here, we need two histograms.
-  Telemetry::Accumulate(Telemetry::WEBRTC_CALL_COUNT_2, 1);
+  /* STUB */
 }
 #endif
 
diff --git a/media/webrtc/signaling/src/peerconnection/WebrtcGlobalInformation.cpp b/media/webrtc/signaling/src/peerconnection/WebrtcGlobalInformation.cpp
index 96bdd5b70..f283d6111 100644
--- a/media/webrtc/signaling/src/peerconnection/WebrtcGlobalInformation.cpp
+++ b/media/webrtc/signaling/src/peerconnection/WebrtcGlobalInformation.cpp
@@ -1194,18 +1194,8 @@ static void GetStatsForLongTermStorage_s(
     rate_limit_bit_pattern |= 2;
   }
 
-  if (query->failed) {
-    Telemetry::Accumulate(
-        Telemetry::WEBRTC_STUN_RATE_LIMIT_EXCEEDED_BY_TYPE_GIVEN_FAILURE,
-        rate_limit_bit_pattern);
-  } else {
-    Telemetry::Accumulate(
-        Telemetry::WEBRTC_STUN_RATE_LIMIT_EXCEEDED_BY_TYPE_GIVEN_SUCCESS,
-        rate_limit_bit_pattern);
-  }
-
-  // Even if Telemetry::Accumulate is threadsafe, we still need to send the
-  // query back to main, since that is where it must be destroyed.
+  // We still need to send the query back to main, since that is where
+  // it must be destroyed.
   NS_DispatchToMainThread(
       WrapRunnableNM(
           &StoreLongTermICEStatisticsImpl_m,
@@ -1216,8 +1206,6 @@ static void GetStatsForLongTermStorage_s(
 
 void WebrtcGlobalInformation::StoreLongTermICEStatistics(
     PeerConnectionImpl& aPc) {
-  Telemetry::Accumulate(Telemetry::WEBRTC_ICE_FINAL_CONNECTION_STATE,
-                        static_cast<uint32_t>(aPc.IceConnectionState()));
 
   if (aPc.IceConnectionState() == PCImplIceConnectionState::New) {
     // ICE has not started; we won't have any remote candidates, so recording