Merge mozilla-central to autoland on a CLOSED TREE

dothq · Oct 2, 2023 · e51bf33 · e51bf33
2 parents 7983ab7 + 37d9d6f
commit e51bf33
Show file tree

Hide file tree

Showing 417 changed files with 14,010 additions and 12,992 deletions.
diff --git a/CLOBBER b/CLOBBER
@@ -22,4 +22,4 @@
 # changes to stick? As of bug 928195, this shouldn't be necessary! Please
 # don't change CLOBBER for WebIDL changes any more.
 
-Modified build files in third_party/libwebrtc - Bug 1832465 - remove libXtst usage from libwebrtc
+Modified build files in third_party/libwebrtc - Bug 1851693 (MOZ) - Moved flexfec_03_header_reader_writer.cc back under non-unified sources to fix build after rename from upstream change ade07ca45e.
diff --git a/dom/media/systemservices/android_video_capture/video_capture_android.cc b/dom/media/systemservices/android_video_capture/video_capture_android.cc
@@ -169,6 +169,7 @@ int32_t VideoCaptureAndroid::Init(const char* deviceUniqueIdUTF8) {
   const int nameLength = strlen(deviceUniqueIdUTF8);
   if (nameLength >= kVideoCaptureUniqueNameLength) return -1;
 
+  RTC_DCHECK_RUN_ON(&api_checker_);
   // Store the device name
   RTC_LOG(LS_INFO) << "VideoCaptureAndroid::Init: " << deviceUniqueIdUTF8;
   _deviceUniqueId = new char[nameLength + 1];
@@ -203,6 +204,7 @@ int32_t VideoCaptureAndroid::StartCapture(
   int min_mfps = 0;
   int max_mfps = 0;
   {
+    RTC_DCHECK_RUN_ON(&api_checker_);
     MutexLock lock(&api_lock_);
 
     if (_deviceInfo.GetBestMatchedCapability(_deviceUniqueId, capability,
@@ -229,6 +231,7 @@ int32_t VideoCaptureAndroid::StartCapture(
   bool started = env->CallBooleanMethod(_jCapturer, j_start, width, height,
                                         min_mfps, max_mfps, j_this);
   if (started) {
+    RTC_DCHECK_RUN_ON(&api_checker_);
     MutexLock lock(&api_lock_);
     _requestedCapability = capability;
     _captureStarted = true;
@@ -240,6 +243,7 @@ int32_t VideoCaptureAndroid::StopCapture() {
   AttachThreadScoped ats(g_jvm_capture);
   JNIEnv* env = ats.env();
   {
+    RTC_DCHECK_RUN_ON(&api_checker_);
     MutexLock lock(&api_lock_);
 
     memset(&_requestedCapability, 0, sizeof(_requestedCapability));
@@ -261,6 +265,7 @@ bool VideoCaptureAndroid::CaptureStarted() {
 }
 
 int32_t VideoCaptureAndroid::CaptureSettings(VideoCaptureCapability& settings) {
+  RTC_DCHECK_RUN_ON(&api_checker_);
   MutexLock lock(&api_lock_);
   settings = _requestedCapability;
   return 0;

diff --git a/dom/media/systemservices/objc_video_capture/video_capture.mm b/dom/media/systemservices/objc_video_capture/video_capture.mm
@@ -57,6 +57,7 @@
   const int32_t name_length = strlen(deviceUniqueIdUTF8);
   if (name_length >= kVideoCaptureUniqueNameLength) return nullptr;
 
+  RTC_DCHECK_RUN_ON(&capture_module->api_checker_);
   capture_module->_deviceUniqueId = new char[name_length + 1];
   strncpy(capture_module->_deviceUniqueId, deviceUniqueIdUTF8, name_length + 1);
   capture_module->_deviceUniqueId[name_length] = '\0';

diff --git a/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp b/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp
@@ -342,7 +342,7 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal(
                 return;
               }
 
-              if (!audioStats->last_packet_received_timestamp_ms) {
+              if (!audioStats->last_packet_received.has_value()) {
                 // By spec: "The lifetime of all RTP monitored objects starts
                 // when the RTP stream is first used: When the first RTP packet
                 // is sent or received on the SSRC it represents"
@@ -400,12 +400,12 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal(
               local.mConcealedSamples.Construct(audioStats->concealed_samples);
               local.mSilentConcealedSamples.Construct(
                   audioStats->silent_concealed_samples);
-              if (audioStats->last_packet_received_timestamp_ms) {
+              if (audioStats->last_packet_received.has_value()) {
                 local.mLastPacketReceivedTimestamp.Construct(
                     RTCStatsTimestamp::FromNtp(
                         aConduit->GetTimestampMaker(),
                         webrtc::Timestamp::Millis(
-                            *audioStats->last_packet_received_timestamp_ms) +
+                            audioStats->last_packet_received->ms()) +
                             webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970))
                         .ToDom());
               }
@@ -444,7 +444,7 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal(
                 return;
               }
 
-              if (!videoStats->rtp_stats.last_packet_received_timestamp_ms) {
+              if (!videoStats->rtp_stats.last_packet_received.has_value()) {
                 // By spec: "The lifetime of all RTP monitored objects starts
                 // when the RTP stream is first used: When the first RTP packet
                 // is sent or received on the SSRC it represents"
@@ -510,7 +510,7 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal(
                   videoStats->frame_counts.key_frames +
                   videoStats->frame_counts.delta_frames);
               local.mJitterBufferDelay.Construct(
-                  videoStats->jitter_buffer_delay_seconds);
+                  videoStats->jitter_buffer_delay.seconds<double>());
               local.mJitterBufferEmittedCount.Construct(
                   videoStats->jitter_buffer_emitted_count);
 
@@ -523,13 +523,12 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal(
                   videoStats->total_inter_frame_delay);
               local.mTotalSquaredInterFrameDelay.Construct(
                   videoStats->total_squared_inter_frame_delay);
-              if (videoStats->rtp_stats.last_packet_received_timestamp_ms) {
+              if (videoStats->rtp_stats.last_packet_received.has_value()) {
                 local.mLastPacketReceivedTimestamp.Construct(
                     RTCStatsTimestamp::FromNtp(
                         aConduit->GetTimestampMaker(),
                         webrtc::Timestamp::Millis(
-                            *videoStats->rtp_stats
-                                 .last_packet_received_timestamp_ms) +
+                            videoStats->rtp_stats.last_packet_received->ms()) +
                             webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970))
                         .ToDom());
               }

diff --git a/dom/media/webrtc/jsapi/RTCRtpSender.cpp b/dom/media/webrtc/jsapi/RTCRtpSender.cpp
@@ -434,7 +434,8 @@ nsTArray<RefPtr<dom::RTCStatsPromise>> RTCRtpSender::GetStatsInternal(
                   }
                 });
 
-            if (streamStats->rtp_stats.first_packet_time_ms == -1) {
+            if (streamStats->rtp_stats.first_packet_time ==
+                webrtc::Timestamp::PlusInfinity()) {
               return;
             }
 

diff --git a/dom/media/webrtc/tests/mochitests/stats.js b/dom/media/webrtc/tests/mochitests/stats.js
@@ -586,7 +586,7 @@ function pedanticChecks(report) {
       let avgJitterBufferDelay =
         stat.jitterBufferDelay / stat.jitterBufferEmittedCount;
       ok(
-        avgJitterBufferDelay > 0.01 && avgJitterBufferDelay < 10,
+        avgJitterBufferDelay > 0.001 && avgJitterBufferDelay < 10,
         `${stat.type}.jitterBufferDelay is a sane number for a short ` +
           `${stat.kind} test. value=${stat.jitterBufferDelay}/${stat.jitterBufferEmittedCount}=${avgJitterBufferDelay}`
       );

diff --git a/dom/media/webrtc/third_party_build/default_config_env b/dom/media/webrtc/third_party_build/default_config_env
@@ -5,41 +5,41 @@
 export MOZ_LIBWEBRTC_SRC=$STATE_DIR/moz-libwebrtc
 
 # The previous fast-forward bug number is used for some error messaging.
-export MOZ_PRIOR_FASTFORWARD_BUG="1839451"
+export MOZ_PRIOR_FASTFORWARD_BUG="1847074"
 
 # Fast-forwarding each Chromium version of libwebrtc should be done
 # under a separate bugzilla bug.  This bug number is used when crafting
 # the commit summary as each upstream commit is vendored into the
 # mercurial repository.  The bug used for the v106 fast-forward was
 # 1800920.
-export MOZ_FASTFORWARD_BUG="1847074"
+export MOZ_FASTFORWARD_BUG="1851693"
 
 # MOZ_NEXT_LIBWEBRTC_MILESTONE and MOZ_NEXT_FIREFOX_REL_TARGET are
 # not used during fast-forward processing, but facilitate generating this
 # default config.  To generate an default config for the next update, run
 # bash dom/media/webrtc/third_party_build/update_default_config_env.sh
-export MOZ_NEXT_LIBWEBRTC_MILESTONE=115
-export MOZ_NEXT_FIREFOX_REL_TARGET=119
+export MOZ_NEXT_LIBWEBRTC_MILESTONE=116
+export MOZ_NEXT_FIREFOX_REL_TARGET=120
 
 # For Chromium release branches, see:
 # https://chromiumdash.appspot.com/branches
 
-# Chromium's v114 release branch was 5735.  This is used to pre-stack
+# Chromium's v115 release branch was 5790.  This is used to pre-stack
 # the previous release branch's commits onto the appropriate base commit
 # (the first common commit between trunk and the release branch).
-export MOZ_PRIOR_UPSTREAM_BRANCH_HEAD_NUM="5735"
+export MOZ_PRIOR_UPSTREAM_BRANCH_HEAD_NUM="5790"
 
-# New target release branch for v115 is branch-heads/5790.  This is used
+# New target release branch for v116 is branch-heads/5845.  This is used
 # to calculate the next upstream commit.
-export MOZ_TARGET_UPSTREAM_BRANCH_HEAD="branch-heads/5790"
+export MOZ_TARGET_UPSTREAM_BRANCH_HEAD="branch-heads/5845"
 
 # For local development 'mozpatches' is fine for a branch name, but when
 # pushing the patch stack to github, it should be named something like
-# 'moz-mods-chr115-for-rel119'.
+# 'moz-mods-chr116-for-rel120'.
 export MOZ_LIBWEBRTC_BRANCH="mozpatches"
 
 # After elm has been merged to mozilla-central, the patch stack in
 # moz-libwebrtc should be pushed to github.  The script
 # push_official_branch.sh uses this branch name when pushing to the
 # public repo.
-export MOZ_LIBWEBRTC_OFFICIAL_BRANCH="moz-mods-chr115-for-rel119"
+export MOZ_LIBWEBRTC_OFFICIAL_BRANCH="moz-mods-chr116-for-rel120"
diff --git a/dom/media/webrtc/third_party_build/gn-configs/webrtc.json b/dom/media/webrtc/third_party_build/gn-configs/webrtc.json
@@ -56,7 +56,7 @@
     "third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc",
     "third_party/libwebrtc/modules/pacing/prioritized_packet_queue.cc",
     "third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc",
-    "third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer2.cc",
+    "third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_03_header_reader_writer.cc",
     "third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_receiver.cc",
     "third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc",
     "third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc",

diff --git a/third_party/libwebrtc/.vpython3 b/third_party/libwebrtc/.vpython3
@@ -83,8 +83,28 @@ wheel: <
   version: "version:3.20.0"
 >
 wheel: <
-  name: "infra/python/wheels/requests-py2_py3"
-  version: "version:2.13.0"
+  name: "infra/python/wheels/requests-py3"
+  version: "version:2.31.0"
+>
+wheel: <
+  name: "infra/python/wheels/idna-py2_py3"
+  version: "version:2.8"
+>
+wheel: <
+  name: "infra/python/wheels/urllib3-py2_py3"
+  version: "version:1.26.6"
+>
+wheel: <
+  name: "infra/python/wheels/certifi-py2_py3"
+  version: "version:2020.11.8"
+>
+wheel: <
+  name: "infra/python/wheels/charset_normalizer-py3"
+  version: "version:2.0.4"
+>
+wheel: <
+  name: "infra/python/wheels/brotli/${vpython_platform}"
+  version: "version:1.0.9"
 >
 
 # Used by:

diff --git a/third_party/libwebrtc/AUTHORS b/third_party/libwebrtc/AUTHORS
@@ -105,6 +105,7 @@ Robert Bares <[email protected]>
 Robert Mader <[email protected]>
 Robert Nagy <[email protected]>
 Ryan Yoakum <[email protected]>
+Samuel Attard <[email protected]>
 Sarah Thompson <[email protected]>
 Satender Saroha <[email protected]>
 Saul Kravitz <[email protected]>

diff --git a/third_party/libwebrtc/BUILD.gn b/third_party/libwebrtc/BUILD.gn
@@ -102,17 +102,13 @@ if (!build_with_chromium && !build_with_mozilla) {
       }
       if (rtc_enable_protobuf) {
         deps += [
-          "audio:low_bandwidth_audio_perf_test",
           "logging:rtc_event_log_rtp_dump",
           "tools_webrtc/perf:webrtc_dashboard_upload",
         ]
       }
       if ((is_linux || is_chromeos) && rtc_use_pipewire) {
         deps += [ "modules/desktop_capture:shared_screencast_stream_test" ]
       }
-      if (is_fuchsia) {
-        deps += [ ":fuchsia_perf_tests" ]
-      }
     }
     if (target_os == "android") {
       deps += [ "tools_webrtc:binary_version_check" ]
@@ -784,24 +780,6 @@ if (rtc_include_tests && !build_with_chromium) {
     }
   }
 
-  if (is_fuchsia) {
-    rtc_test("fuchsia_perf_tests") {
-      testonly = true
-      deps = [
-        #TODO(fxbug.dev/115601) - Enable when fixed
-        "call:call_perf_tests",
-        "modules/audio_coding:audio_coding_perf_tests",
-        "modules/audio_processing:audio_processing_perf_tests",
-        "pc:peerconnection_perf_tests",
-        "test:test_main",
-        "video:video_full_stack_tests",
-        "video:video_pc_full_stack_tests",
-      ]
-
-      data = webrtc_perf_tests_resources
-    }
-  }
-
   rtc_test("webrtc_nonparallel_tests") {
     testonly = true
     deps = [ "rtc_base:rtc_base_nonparallel_tests" ]
-Original file line number
+Diff line change
@@ Expand Up @@
                       }
                     });
-                if (streamStats->rtp_stats.first_packet_time_ms == -1) {
+                if (streamStats->rtp_stats.first_packet_time ==
+                    webrtc::Timestamp::PlusInfinity()) {
                   return;
                 }
@@ Expand Down @@