Update iOS version from 17 to 18.

Change-Id: Id4b782f6f10dca3e5cc7f34c2f6620253b3f24d0 Bug: None Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378042 Reviewed-by: Christoffer Dewerin <jansson@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43949}
Roll chromium_revision c50f9710da..40885069c1 (1422420:1422971)
2025-02-21 01:42:29 -08:00 · 2025-02-21 00:37:55 -08:00 · 2025-02-21 00:01:13 -08:00 · 2025-02-20 23:50:12 -08:00 · 2025-02-20 21:44:16 -08:00 · 2025-02-20 08:48:47 -08:00
59 changed files with 206 additions and 152 deletions
--- a/34
+++ b/34
@ -10,7 +10,7 @@ vars = {
  # chromium waterfalls. More info at: crbug.com/570091.
  'checkout_configuration': 'default',
  'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
-  'chromium_revision': 'c50f9710da99968a7817f069dafba47c5155cdcd',
+  'chromium_revision': '40885069c19986b2dd96326b6e8dddd99c5f8ac4',

  # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
  # process the raw profiles produced by instrumented targets (built with
@ -64,24 +64,24 @@ vars = {
 deps = {
  # TODO(kjellander): Move this to be Android-only.
  'src/base':
-    'https://chromium.googlesource.com/chromium/src/base@9e81d0aac944aea8355604ddd28065af1d8a24e3',
+    'https://chromium.googlesource.com/chromium/src/base@a62135f7fa7fb9342b98cabc1654b385888c52f8',
  'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@e42ba23e14b15bbfc89a809dee1c8f700d80c736',
+    'https://chromium.googlesource.com/chromium/src/build@530d40d7d32fdb762b8eec27d0b30325eeca2dfe',
  'src/buildtools':
-    'https://chromium.googlesource.com/chromium/src/buildtools@d1ed719a33042a69290036402227949ec4d11877',
+    'https://chromium.googlesource.com/chromium/src/buildtools@99bd0872895b0e7da5a5fa241b24b5b58f508641',
  # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
  'src/examples/androidtests/third_party/gradle': {
    'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3',
    'condition': 'checkout_android',
  },
  'src/ios': {
-    'url': 'https://chromium.googlesource.com/chromium/src/ios@f497e652b6cd0bfd75094548dd9bfe872cba7c30',
+    'url': 'https://chromium.googlesource.com/chromium/src/ios@f8fdd3967c7182ced94963b65fed98b4b77c824c',
    'condition': 'checkout_ios',
  },
  'src/testing':
-    'https://chromium.googlesource.com/chromium/src/testing@75065cfcbb2b3db725f614f0d97bafcaaad3bb4b',
+    'https://chromium.googlesource.com/chromium/src/testing@0d1f0f2d5d1eec0d377f81f5c1dca82527ca13d3',
  'src/third_party':
-    'https://chromium.googlesource.com/chromium/src/third_party@f004cc0a39eb7c5125a84f931ebfe1fadf8a4458',
+    'https://chromium.googlesource.com/chromium/src/third_party@adf98b9a1ee057cb1cf57e3fdc39ca91309627c4',

  'src/buildtools/linux64': {
    'packages': [
@ -164,11 +164,11 @@ deps = {
  'src/third_party/clang-format/script':
    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2',
  'src/third_party/libc++/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@e36be6784f33c592ecc89da2a829807920d374ef',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@1a88911caa16345beb23d141f0361d58f3c4fcf5',
  'src/third_party/libc++abi/src':
    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@94c5d7a8edc09f0680aee57548c0b5d400c2840d',
  'src/third_party/llvm-libc/src':
-    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@99782c3117bf68302f6d9080d70ac5bd86fa72f6',
+    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@e722dac269c4c938fcd197d0e37987eb29cb2989',
  'src/third_party/libunwind/src':
    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@62e217a12ee1133833d9890b2f7adde900e4efbd',

@ -286,11 +286,11 @@ deps = {
  },

  'src/third_party/boringssl/src':
-    'https://boringssl.googlesource.com/boringssl.git@294ab9730c570213b496cfc2fc14b3c0bfcd4bcc',
+    'https://boringssl.googlesource.com/boringssl.git@0c59e9e55bdaadadf49f3c092b097e72cd4b39cc',
  'src/third_party/breakpad/breakpad':
    'https://chromium.googlesource.com/breakpad/breakpad.git@0dfd77492fdb0dcd06027c5842095e2e908adc90',
  'src/third_party/catapult':
-    'https://chromium.googlesource.com/catapult.git@30345af7ee8bf6e5ac59d4fc9a2fa3edbb707a4a',
+    'https://chromium.googlesource.com/catapult.git@09f963b5f4b4304a0f1b9e75fc610de2eb0603ef',
  'src/third_party/ced/src': {
    'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
  },
@ -303,7 +303,7 @@ deps = {
  'src/third_party/crc32c/src':
    'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
  'src/third_party/depot_tools':
-    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@ab387d1ddc530d405f662904f2df292f700e4b0f',
+    'https://chromium.googlesource.com/chromium/tools/depot_tools.git@2259aa176628b8c8ef67a1e1c6dab94cb69ba3c9',
  'src/third_party/ffmpeg':
    'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@fbce2a76c00cd2e5aeffe3c2e71d44c284ec52d6',
  'src/third_party/flatbuffers/src':
@ -389,7 +389,7 @@ deps = {
  'src/third_party/libFuzzer/src':
    'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@e31b99917861f891308269c36a32363b120126bb',
  'src/third_party/fuzztest/src':
-    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@1d61720657533c73cbffda1e909e2130fcae5029',
+    'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@409e5199f68ac4aafd4b1e9aeae0bb79e464b94a',
  'src/third_party/libjpeg_turbo':
    'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@927aabfcd26897abb9776ecf2a6c38ea5bb52ab6',
  'src/third_party/libsrtp':
@ -403,7 +403,7 @@ deps = {
      'condition': 'checkout_android',
  },
  'src/third_party/perfetto':
-    'https://android.googlesource.com/platform/external/perfetto.git@1e97ffd1534d3cff6b33ba2c01ba30eff2f249cb',
+    'https://android.googlesource.com/platform/external/perfetto.git@072b48f0e00550ae76cde0d21b5e5793c2613443',
  'src/third_party/protobuf-javascript/src':
    Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + '426b2e025aa59c23a1b6e19fbacd6b4a11bfad16',
  'src/third_party/libvpx/source/libvpx':
@ -438,7 +438,7 @@ deps = {
      'packages': [
          {
              'package': 'chromium/third_party/r8',
-              'version': 'QDwsH_fhjzc4w9ejy5IKJrAl9g6ct71e1UJiFwGoqYMC',
+              'version': '2ChXGrcBJ9WNhFhS069j3L6VMp0jw5xyyPqKhlOqj_sC',
          },
      ],
      'condition': 'checkout_android',
@ -462,7 +462,7 @@ deps = {
    'condition': 'checkout_android',
  },
  'src/tools':
-    'https://chromium.googlesource.com/chromium/src/tools@367e3830749cae49cfaf9646b5e8c99270679b1d',
+    'https://chromium.googlesource.com/chromium/src/tools@a4cad1e12adcdccf32335806ce3c73fc37650d84',

  'src/third_party/espresso': {
      'packages': [
@ -501,7 +501,7 @@ deps = {
    'packages': [
      {
          'package': 'chromium/third_party/androidx',
-          'version': 'R1wNTeO5kH2bzp1NW4g87DLWtiDXaImAQJplHSR-108C',
+          'version': '9i2Jdh3nRH-lCfGUTvpGecSiz9lyJ3mGDYTNz7N04csC',
      },
    ],
    'condition': 'checkout_android and non_git_source',
--- a/api/audio/echo_canceller3_config.cc
+++ b/api/audio/echo_canceller3_config.cc
@ -18,7 +18,7 @@
 namespace webrtc {
 namespace {
 bool Limit(float* value, float min, float max) {
-  float clamped = rtc::SafeClamp(*value, min, max);
+  float clamped = SafeClamp(*value, min, max);
  clamped = std::isfinite(clamped) ? clamped : min;
  bool res = *value == clamped;
  *value = clamped;
@ -26,14 +26,14 @@ bool Limit(float* value, float min, float max) {
 }

 bool Limit(size_t* value, size_t min, size_t max) {
-  size_t clamped = rtc::SafeClamp(*value, min, max);
+  size_t clamped = SafeClamp(*value, min, max);
  bool res = *value == clamped;
  *value = clamped;
  return res;
 }

 bool Limit(int* value, int min, int max) {
-  int clamped = rtc::SafeClamp(*value, min, max);
+  int clamped = SafeClamp(*value, min, max);
  bool res = *value == clamped;
  *value = clamped;
  return res;
--- a/api/audio_codecs/L16/audio_encoder_L16.cc
+++ b/api/audio_codecs/L16/audio_encoder_L16.cc
@ -45,7 +45,7 @@ std::optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
  if (ptime_iter != format.parameters.end()) {
    const auto ptime = StringToNumber<int>(ptime_iter->second);
    if (ptime && *ptime > 0) {
-      config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
+      config.frame_size_ms = SafeClamp(10 * (*ptime / 10), 10, 60);
    }
  }
  if (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) {
--- a/api/audio_codecs/g711/audio_encoder_g711.cc
+++ b/api/audio_codecs/g711/audio_encoder_g711.cc
@ -46,7 +46,7 @@ std::optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
    if (ptime_iter != format.parameters.end()) {
      const auto ptime = StringToNumber<int>(ptime_iter->second);
      if (ptime && *ptime > 0) {
-        config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
+        config.frame_size_ms = SafeClamp(10 * (*ptime / 10), 10, 60);
      }
    }
    if (!config.IsOk()) {
--- a/api/audio_codecs/g722/audio_encoder_g722.cc
+++ b/api/audio_codecs/g722/audio_encoder_g722.cc
@ -46,7 +46,7 @@ std::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
    auto ptime = StringToNumber<int>(ptime_iter->second);
    if (ptime && *ptime > 0) {
      const int whole_packets = *ptime / 10;
-      config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 10, 60);
+      config.frame_size_ms = SafeClamp<int>(whole_packets * 10, 10, 60);
    }
  }
  if (!config.IsOk()) {
--- a/audio/audio_state.cc
+++ b/audio/audio_state.cc
@ -141,13 +141,16 @@ void AudioState::SetRecording(bool enabled) {
  RTC_LOG(LS_INFO) << "SetRecording(" << enabled << ")";
  RTC_DCHECK_RUN_ON(&thread_checker_);
  if (recording_enabled_ != enabled) {
+    auto* adm = config_.audio_device_module.get();
    recording_enabled_ = enabled;
    if (enabled) {
      if (!sending_streams_.empty()) {
-        config_.audio_device_module->StartRecording();
+        if (adm->InitRecording() == 0) {
+          adm->StartRecording();
+        }
      }
    } else {
-      config_.audio_device_module->StopRecording();
+      adm->StopRecording();
    }
  }
 }
--- a/audio/audio_state_unittest.cc
+++ b/audio/audio_state_unittest.cc
@ -26,6 +26,7 @@ namespace test {
 namespace {

 using ::testing::_;
+using ::testing::InSequence;
 using ::testing::Matcher;
 using ::testing::NiceMock;
 using ::testing::StrictMock;
@ -357,6 +358,36 @@ TEST_P(AudioStateTest,
      audio_buffer, n_samples_out, &elapsed_time_ms, &ntp_time_ms);
 }

+TEST_P(AudioStateTest, AlwaysCallInitRecordingBeforeStartRecording) {
+  ConfigHelper helper(GetParam());
+  rtc::scoped_refptr<internal::AudioState> audio_state(
+      rtc::make_ref_counted<internal::AudioState>(helper.config()));
+
+  auto* adm = reinterpret_cast<MockAudioDeviceModule*>(
+      helper.config().audio_device_module.get());
+
+  MockAudioSendStream stream;
+  {
+    InSequence s;
+    EXPECT_CALL(*adm, InitRecording());
+    EXPECT_CALL(*adm, StartRecording());
+    audio_state->AddSendingStream(&stream, kSampleRate, kNumberOfChannels);
+  }
+
+  EXPECT_CALL(*adm, StopRecording());
+  audio_state->SetRecording(false);
+
+  {
+    InSequence s;
+    EXPECT_CALL(*adm, InitRecording());
+    EXPECT_CALL(*adm, StartRecording());
+    audio_state->SetRecording(true);
+  }
+
+  EXPECT_CALL(*adm, StopRecording());
+  audio_state->RemoveSendingStream(&stream);
+}
+
 INSTANTIATE_TEST_SUITE_P(AudioStateTest,
                         AudioStateTest,
                         Values(ConfigHelper::Params({false, false}),
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@ -1058,8 +1058,8 @@ bool ChannelReceive::SetMinimumPlayoutDelay(int delay_ms) {
  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
  // Limit to range accepted by both VoE and ACM, so we're at least getting as
  // close as possible, instead of failing.
-  delay_ms = rtc::SafeClamp(delay_ms, kVoiceEngineMinMinPlayoutDelayMs,
-                            kVoiceEngineMaxMinPlayoutDelayMs);
+  delay_ms = SafeClamp(delay_ms, kVoiceEngineMinMinPlayoutDelayMs,
+                       kVoiceEngineMaxMinPlayoutDelayMs);
  if (!neteq_->SetMinimumDelay(delay_ms)) {
    RTC_DLOG(LS_ERROR)
        << "SetMinimumPlayoutDelay() failed to set min playout delay "
--- a/call/bitrate_allocator.cc
+++ b/call/bitrate_allocator.cc
@ -494,7 +494,7 @@ void BitrateAllocator::OnNetworkEstimateChanged(TargetTransferRate msg) {

  int loss_ratio_255 = msg.network_estimate.loss_rate_ratio * 255;
  last_fraction_loss_ =
-      rtc::dchecked_cast<uint8_t>(rtc::SafeClamp(loss_ratio_255, 0, 255));
+      rtc::dchecked_cast<uint8_t>(SafeClamp(loss_ratio_255, 0, 255));
  last_rtt_ = msg.network_estimate.round_trip_time.ms();
  last_bwe_period_ms_ = msg.network_estimate.bwe_period.ms();

--- a/call/receive_time_calculator.cc
+++ b/call/receive_time_calculator.cc
@ -57,7 +57,7 @@ int64_t ReceiveTimeCalculator::ReconcileReceiveTimes(int64_t packet_time_us,
                                                     int64_t safe_time_us) {
  int64_t stall_time_us = system_time_us - packet_time_us;
  if (total_system_time_passed_us_ < config_.stall_threshold->us()) {
-    stall_time_us = rtc::SafeMin(stall_time_us, config_.max_stall->us());
+    stall_time_us = SafeMin(stall_time_us, config_.max_stall->us());
  }
  int64_t corrected_time_us = safe_time_us - stall_time_us;

@ -107,8 +107,8 @@ int64_t ReceiveTimeCalculator::ReconcileReceiveTimes(int64_t packet_time_us,
    if (forward_clock_reset || obvious_backward_clock_reset ||
        small_reset_during_stall_) {
      corrected_time_us = last_corrected_time_us_ +
-                          rtc::SafeClamp(packet_time_delta_us, 0,
-                                         config_.max_packet_time_repair->us());
+                          SafeClamp(packet_time_delta_us, 0,
+                                    config_.max_packet_time_repair->us());
    }
  }

--- a/call/version.cc
+++ b/call/version.cc
@ -13,7 +13,7 @@
 namespace webrtc {

 // The timestamp is always in UTC.
-const char* const kSourceTimestamp = "WebRTC source stamp 2025-02-20T04:02:19";
+const char* const kSourceTimestamp = "WebRTC source stamp 2025-02-21T04:05:41";

 void LoadWebRTCVersionInRegister() {
  // Using volatile to instruct the compiler to not optimize `p` away even
--- a/infra/config/commit-queue.cfg
+++ b/infra/config/commit-queue.cfg
@ -90,6 +90,9 @@ config_groups {
      builders {
        name: "webrtc/try/ios_compile_arm64_rel"
      }
+      builders {
+        name: "webrtc/try/ios_dbg_simulator"
+      }
      builders {
        name: "webrtc/try/linux_asan"
      }
@ -267,6 +270,9 @@ config_groups {
      builders {
        name: "webrtc/try/ios_compile_arm64_rel"
      }
+      builders {
+        name: "webrtc/try/ios_dbg_simulator"
+      }
      builders {
        name: "webrtc/try/linux_asan"
      }
--- a/infra/config/config.star
+++ b/infra/config/config.star
@ -53,7 +53,6 @@ def os_from_name(name):
 # useful when a failure can be safely ignored while fixing it without
 # blocking the LKGR finder on it.
 skipped_lkgr_bots = [
-    "iOS Debug (simulator)",
 ]

 # Use LUCI Scheduler BBv2 names and add Scheduler realms configs.
@ -761,8 +760,7 @@ ios_builder("iOS64 Release", "iOS|arm64|rel")
 ios_try_job("ios_compile_arm64_rel")
 ios_builder("iOS Debug (simulator)", "iOS|x64|sim")

-# TODO: b/396118151 - Reenable the bot when the issue is fixed.
-ios_try_job("ios_dbg_simulator", cq = None)
+ios_try_job("ios_dbg_simulator")
 ios_builder("iOS API Framework Builder", "iOS|fat|size", recipe = "ios_api_framework", prioritized = True)
 ios_try_job("ios_api_framework", recipe = "ios_api_framework")

--- a/infra/config/cr-buildbucket.cfg
+++ b/infra/config/cr-buildbucket.cfg
@ -1907,6 +1907,7 @@ buckets {
        '          "Win64 Debug (Clang)",'
        '          "Win64 Release (Clang)",'
        '          "iOS API Framework Builder",'
+        '          "iOS Debug (simulator)",'
        '          "iOS64 Debug",'
        '          "iOS64 Release"'
        '        ]'
--- a/infra/config/luci-notify.cfg
+++ b/infra/config/luci-notify.cfg
@ -943,6 +943,11 @@ notifiers {
    name: "iOS Debug (simulator)"
    repository: "https://webrtc.googlesource.com/src"
  }
+  tree_closers {
+    tree_status_host: "webrtc-status.appspot.com"
+    failed_step_regexp: "bot_update|compile|gclient runhooks|runhooks|update|extract build|cleanup_temp|taskkill|compile|gn"
+    failed_step_regexp_exclude: ".*\\(experimental\\).*"
+  }
 }
 notifiers {
  notifications {
--- a/infra/specs/internal.client.webrtc.json
+++ b/infra/specs/internal.client.webrtc.json
@ -28,7 +28,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
@ -67,7 +67,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
@ -107,7 +107,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "hard_timeout": 7200,
@ -149,7 +149,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
@ -189,7 +189,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
@ -228,7 +228,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
@ -267,7 +267,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
@ -306,7 +306,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
@ -345,7 +345,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
@ -384,7 +384,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
@ -481,7 +481,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
@ -520,7 +520,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
@ -560,7 +560,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "hard_timeout": 7200,
@ -602,7 +602,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
@ -642,7 +642,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
@ -681,7 +681,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
@ -720,7 +720,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
@ -759,7 +759,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
@ -798,7 +798,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
@ -837,7 +837,7 @@
            }
          ],
          "dimensions": {
-            "os": "iOS-17.6.1",
+            "os": "iOS-18",
            "pool": "chrome.tests"
          },
          "named_caches": [
--- a/infra/specs/mixins.pyl
+++ b/infra/specs/mixins.pyl
@ -66,10 +66,10 @@
      'has_native_resultdb_integration': True
    }
  },
-  'ios-device-17': {
+  'ios-device-18': {
    'swarming': {
      'dimensions': {
-        'os': 'iOS-17.6.1',
+        'os': 'iOS-18',
        'pool': 'chrome.tests'
      }
    }
--- a/infra/specs/mixins_webrtc.pyl
+++ b/infra/specs/mixins_webrtc.pyl
@ -45,10 +45,10 @@
      '--test-arg=--undefok=test_launcher_summary_output'
    ],
  },
-  'ios-device-17': {
+  'ios-device-18': {
    'swarming': {
      'dimensions': {
-        'os': 'iOS-17.6.1',
+        'os': 'iOS-18',
        'pool': 'chrome.tests',
      }
    }
--- a/infra/specs/waterfalls.pyl
+++ b/infra/specs/waterfalls.pyl
@ -322,7 +322,7 @@
    'machines': {
      'iOS64 Debug': {
        'mixins': [
-          'ios-device-17', 'webrtc-xctest', 'chrome-tester-service-account',
+          'ios-device-18', 'webrtc-xctest', 'chrome-tester-service-account',
          'xcode_15_main', 'mac_toolchain', 'has_native_resultdb_integration',
          'out_dir_arg'
        ],
@ -345,7 +345,7 @@
      },
      'iOS64 Release': {
        'mixins': [
-          'ios-device-17', 'webrtc-xctest', 'chrome-tester-service-account',
+          'ios-device-18', 'webrtc-xctest', 'chrome-tester-service-account',
          'xcode_15_main', 'mac_toolchain', 'has_native_resultdb_integration',
          'out_dir_arg'
        ],
--- a/logging/rtc_event_log/rtc_event_log_impl.cc
+++ b/logging/rtc_event_log/rtc_event_log_impl.cc
@ -235,8 +235,8 @@ void RtcEventLogImpl::ScheduleOutput() {
  };
  const int64_t now_ms = rtc::TimeMillis();
  const int64_t time_since_output_ms = now_ms - last_output_ms_;
-  const int32_t delay = rtc::SafeClamp(output_period_ms_ - time_since_output_ms,
-                                       0, output_period_ms_);
+  const int32_t delay =
+      SafeClamp(output_period_ms_ - time_since_output_ms, 0, output_period_ms_);
  task_queue_->PostDelayedTask(std::move(output_task),
                               TimeDelta::Millis(delay));
 }
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@ -733,9 +733,9 @@ void AudioEncoderOpusImpl::SetProjectedPacketLossRate(float fraction) {
 }

 void AudioEncoderOpusImpl::SetTargetBitrate(int bits_per_second) {
-  const int new_bitrate = rtc::SafeClamp<int>(
-      bits_per_second, AudioEncoderOpusConfig::kMinBitrateBps,
-      AudioEncoderOpusConfig::kMaxBitrateBps);
+  const int new_bitrate =
+      SafeClamp<int>(bits_per_second, AudioEncoderOpusConfig::kMinBitrateBps,
+                     AudioEncoderOpusConfig::kMaxBitrateBps);
  if (config_.bitrate_bps && *config_.bitrate_bps != new_bitrate) {
    config_.bitrate_bps = new_bitrate;
    RTC_DCHECK(config_.IsOk());
--- a/modules/audio_coding/neteq/delay_constraints.cc
+++ b/modules/audio_coding/neteq/delay_constraints.cc
@ -100,7 +100,7 @@ void DelayConstraints::UpdateEffectiveMinimumDelay() {
  // Clamp `base_minimum_delay_ms_` into the range which can be effectively
  // used.
  const int base_minimum_delay_ms =
-      rtc::SafeClamp(base_minimum_delay_ms_, 0, MinimumDelayUpperBound());
+      SafeClamp(base_minimum_delay_ms_, 0, MinimumDelayUpperBound());
  effective_minimum_delay_ms_ =
      std::max(minimum_delay_ms_, base_minimum_delay_ms);
 }
--- a/modules/audio_coding/neteq/merge.cc
+++ b/modules/audio_coding/neteq/merge.cc
@ -211,8 +211,8 @@ int16_t Merge::SignalScaling(const int16_t* input,
                             size_t input_length,
                             const int16_t* expanded_signal) const {
  // Adjust muting factor if new vector is more or less of the BGN energy.
-  const auto mod_input_length = rtc::SafeMin<size_t>(
-      64 * rtc::dchecked_cast<size_t>(fs_mult_), input_length);
+  const auto mod_input_length =
+      SafeMin<size_t>(64 * rtc::dchecked_cast<size_t>(fs_mult_), input_length);
  const int16_t expanded_max =
      WebRtcSpl_MaxAbsValueW16(expanded_signal, mod_input_length);
  int32_t factor =
--- a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
@ -566,7 +566,7 @@ TEST_P(AdaptiveFirFilterMultiChannel, FilterAndAdapt) {
                     e.begin(),
                     [&](float a, float b) { return a - b * kScale; });
      std::for_each(e.begin(), e.end(),
-                    [](float& a) { a = rtc::SafeClamp(a, -32768.f, 32767.f); });
+                    [](float& a) { a = SafeClamp(a, -32768.f, 32767.f); });
      fft.ZeroPaddedFft(e, Aec3Fft::Window::kRectangular, &E);
      for (auto& o : output) {
        for (size_t k = 0; k < kBlockSize; ++k) {
--- a/modules/audio_processing/aec3/coarse_filter_update_gain_unittest.cc
+++ b/modules/audio_processing/aec3/coarse_filter_update_gain_unittest.cc
@ -100,7 +100,7 @@ void RunFilterUpdateTest(int num_blocks_to_process,
                   e_coarse.begin(),
                   [&](float a, float b) { return a - b * kScale; });
    std::for_each(e_coarse.begin(), e_coarse.end(),
-                  [](float& a) { a = rtc::SafeClamp(a, -32768.f, 32767.f); });
+                  [](float& a) { a = SafeClamp(a, -32768.f, 32767.f); });
    fft.ZeroPaddedFft(e_coarse, Aec3Fft::Window::kRectangular, &E_coarse);

    std::array<float, kFftLengthBy2Plus1> render_power;
--- a/modules/audio_processing/aec3/echo_remover_metrics.cc
+++ b/modules/audio_processing/aec3/echo_remover_metrics.cc
@ -149,7 +149,7 @@ int TransformDbMetricForReporting(bool negate,
  if (negate) {
    new_value = -new_value;
  }
-  return static_cast<int>(rtc::SafeClamp(new_value, min_value, max_value));
+  return static_cast<int>(SafeClamp(new_value, min_value, max_value));
 }

 }  // namespace aec3
--- a/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc
+++ b/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc
@ -162,7 +162,7 @@ void RunFilterUpdateTest(const Environment& env,
                   e_refined.begin(),
                   [&](float a, float b) { return a - b * kScale; });
    std::for_each(e_refined.begin(), e_refined.end(),
-                  [](float& a) { a = rtc::SafeClamp(a, -32768.f, 32767.f); });
+                  [](float& a) { a = SafeClamp(a, -32768.f, 32767.f); });
    fft.ZeroPaddedFft(e_refined, Aec3Fft::Window::kRectangular, &E_refined);
    for (size_t k = 0; k < kBlockSize; ++k) {
      s[k] = kScale * s_scratch[k + kFftLengthBy2];
@ -175,7 +175,7 @@ void RunFilterUpdateTest(const Environment& env,
                   e_coarse.begin(),
                   [&](float a, float b) { return a - b * kScale; });
    std::for_each(e_coarse.begin(), e_coarse.end(),
-                  [](float& a) { a = rtc::SafeClamp(a, -32768.f, 32767.f); });
+                  [](float& a) { a = SafeClamp(a, -32768.f, 32767.f); });
    fft.ZeroPaddedFft(e_coarse, Aec3Fft::Window::kRectangular, &E_coarse);

    // Compute spectra for future use.
--- a/modules/audio_processing/aec3/signal_dependent_erle_estimator.cc
+++ b/modules/audio_processing/aec3/signal_dependent_erle_estimator.cc
@ -205,12 +205,12 @@ void SignalDependentErleEstimator::Update(
      float correction_factor =
          correction_factors_[ch][n_active_sections_[ch][k]]
                             [band_to_subband_[k]];
-      erle_[ch][k] = rtc::SafeClamp(average_erle[ch][k] * correction_factor,
-                                    min_erle_, max_erle_[band_to_subband_[k]]);
+      erle_[ch][k] = SafeClamp(average_erle[ch][k] * correction_factor,
+                               min_erle_, max_erle_[band_to_subband_[k]]);
      if (use_onset_detection_) {
-        erle_onset_compensated_[ch][k] = rtc::SafeClamp(
-            average_erle_onset_compensated[ch][k] * correction_factor,
-            min_erle_, max_erle_[band_to_subband_[k]]);
+        erle_onset_compensated_[ch][k] =
+            SafeClamp(average_erle_onset_compensated[ch][k] * correction_factor,
+                      min_erle_, max_erle_[band_to_subband_[k]]);
      }
    }
  }
@ -306,7 +306,7 @@ void SignalDependentErleEstimator::UpdateCorrectionFactors(
        alpha = static_cast<float>(is_erle_updated[subband]) * alpha;
        erle_estimators_[ch][idx][subband] +=
            alpha * (new_erle[subband] - erle_estimators_[ch][idx][subband]);
-        erle_estimators_[ch][idx][subband] = rtc::SafeClamp(
+        erle_estimators_[ch][idx][subband] = SafeClamp(
            erle_estimators_[ch][idx][subband], min_erle_, max_erle_[subband]);
      }

@ -317,8 +317,8 @@ void SignalDependentErleEstimator::UpdateCorrectionFactors(
        alpha = static_cast<float>(is_erle_updated[subband]) * alpha;
        erle_ref_[ch][subband] +=
            alpha * (new_erle[subband] - erle_ref_[ch][subband]);
-        erle_ref_[ch][subband] = rtc::SafeClamp(erle_ref_[ch][subband],
-                                                min_erle_, max_erle_[subband]);
+        erle_ref_[ch][subband] =
+            SafeClamp(erle_ref_[ch][subband], min_erle_, max_erle_[subband]);
      }

      for (size_t subband = 0; subband < kSubbands; ++subband) {
--- a/modules/audio_processing/aec3/subband_erle_estimator.cc
+++ b/modules/audio_processing/aec3/subband_erle_estimator.cc
@ -141,7 +141,7 @@ void SubbandErleEstimator::UpdateBands(
            if (!use_min_erle_during_onsets_) {
              float alpha =
                  new_erle[k] < erle_during_onsets_[ch][k] ? 0.3f : 0.15f;
-              erle_during_onsets_[ch][k] = rtc::SafeClamp(
+              erle_during_onsets_[ch][k] = SafeClamp(
                  erle_during_onsets_[ch][k] +
                      alpha * (new_erle[k] - erle_during_onsets_[ch][k]),
                  min_erle_, max_erle_[k]);
@ -159,8 +159,7 @@ void SubbandErleEstimator::UpdateBands(
      if (new_erle < erle) {
        alpha = low_render_energy ? 0.f : 0.1f;
      }
-      erle =
-          rtc::SafeClamp(erle + alpha * (new_erle - erle), min_erle, max_erle);
+      erle = SafeClamp(erle + alpha * (new_erle - erle), min_erle, max_erle);
    };

    for (size_t k = 1; k < kFftLengthBy2; ++k) {
--- a/modules/audio_processing/aec3/suppression_filter.cc
+++ b/modules/audio_processing/aec3/suppression_filter.cc
@ -171,7 +171,7 @@ void SuppressionFilter::ApplyGain(
    for (int b = 0; b < e->NumBands(); ++b) {
      auto e_band = e->View(b, ch);
      for (size_t i = 0; i < kFftLengthBy2; ++i) {
-        e_band[i] = rtc::SafeClamp(e_band[i], -32768.f, 32767.f);
+        e_band[i] = SafeClamp(e_band[i], -32768.f, 32767.f);
      }
    }
  }
--- a/modules/audio_processing/agc/agc_manager_direct.cc
+++ b/modules/audio_processing/agc/agc_manager_direct.cc
@ -152,7 +152,7 @@ int GetSpeechLevelErrorDb(float speech_level_dbfs, float speech_probability) {
    return 0;
  }

-  const float speech_level = rtc::SafeClamp<float>(
+  const float speech_level = SafeClamp<float>(
      speech_level_dbfs, kMinSpeechLevelDbfs, kMaxSpeechLevelDbfs);

  return std::round(kOverrideTargetSpeechLevelDbfs - speech_level);
@ -376,7 +376,7 @@ void MonoAgc::UpdateGain(int rms_error_db) {

  // Handle as much error as possible with the compressor first.
  int raw_compression =
-      rtc::SafeClamp(rms_error, kMinCompressionGain, max_compression_gain_);
+      SafeClamp(rms_error, kMinCompressionGain, max_compression_gain_);

  // Deemphasize the compression gain error. Move halfway between the current
  // target and the newly received target. This serves to soften perceptible
@ -397,8 +397,8 @@ void MonoAgc::UpdateGain(int rms_error_db) {
  // raw rather than deemphasized compression here as we would otherwise
  // shrink the amount of slack the compressor provides.
  const int residual_gain =
-      rtc::SafeClamp(rms_error - raw_compression, -kMaxResidualGainChange,
-                     kMaxResidualGainChange);
+      SafeClamp(rms_error - raw_compression, -kMaxResidualGainChange,
+                kMaxResidualGainChange);
  RTC_DLOG(LS_INFO) << "[agc] rms_error=" << rms_error
                    << ", target_compression=" << target_compression_
                    << ", residual_gain=" << residual_gain;
--- a/modules/audio_processing/agc/agc_manager_direct_unittest.cc
+++ b/modules/audio_processing/agc/agc_manager_direct_unittest.cc
@ -258,8 +258,8 @@ class SpeechSamplesReader {
      // Apply gain and copy samples into `audio_buffer_`.
      std::transform(buffer_.begin(), buffer_.end(),
                     audio_buffer_.channels()[0], [gain](int16_t v) -> float {
-                       return rtc::SafeClamp(static_cast<float>(v) * gain,
-                                             kMinSample, kMaxSample);
+                       return SafeClamp(static_cast<float>(v) * gain,
+                                        kMinSample, kMaxSample);
                     });

      agc.AnalyzePreProcess(audio_buffer_);
@ -292,8 +292,8 @@ class SpeechSamplesReader {
      // Apply gain and copy samples into `audio_buffer_`.
      std::transform(buffer_.begin(), buffer_.end(),
                     audio_buffer_.channels()[0], [gain](int16_t v) -> float {
-                       return rtc::SafeClamp(static_cast<float>(v) * gain,
-                                             kMinSample, kMaxSample);
+                       return SafeClamp(static_cast<float>(v) * gain,
+                                        kMinSample, kMaxSample);
                     });

      agc.AnalyzePreProcess(audio_buffer_);
--- a/modules/audio_processing/agc2/adaptive_digital_gain_controller.cc
+++ b/modules/audio_processing/agc2/adaptive_digital_gain_controller.cc
@ -96,8 +96,8 @@ float ComputeGainChangeThisFrameDb(float target_gain_db,
  if (!gain_increase_allowed) {
    target_gain_difference_db = std::min(target_gain_difference_db, 0.0f);
  }
-  return rtc::SafeClamp(target_gain_difference_db, -max_gain_decrease_db,
-                        max_gain_increase_db);
+  return SafeClamp(target_gain_difference_db, -max_gain_decrease_db,
+                   max_gain_increase_db);
 }

 }  // namespace
--- a/modules/audio_processing/agc2/clipping_predictor.cc
+++ b/modules/audio_processing/agc2/clipping_predictor.cc
@ -151,7 +151,7 @@ class ClippingEventPredictor : public ClippingPredictor {
    }
    if (PredictClippingEvent(channel)) {
      const int new_level =
-          rtc::SafeClamp(level - default_step, min_mic_level, max_mic_level);
+          SafeClamp(level - default_step, min_mic_level, max_mic_level);
      const int step = level - new_level;
      if (step > 0) {
        return step;
@ -296,15 +296,15 @@ class ClippingPeakPredictor : public ClippingPredictor {
        step = default_step;
      } else {
        const int estimated_gain_change =
-            rtc::SafeClamp(-static_cast<int>(std::ceil(estimate_db.value())),
-                           -kClippingPredictorMaxGainChange, 0);
+            SafeClamp(-static_cast<int>(std::ceil(estimate_db.value())),
+                      -kClippingPredictorMaxGainChange, 0);
        step =
            std::max(level - ComputeVolumeUpdate(estimated_gain_change, level,
                                                 min_mic_level, max_mic_level),
                     default_step);
      }
      const int new_level =
-          rtc::SafeClamp(level - step, min_mic_level, max_mic_level);
+          SafeClamp(level - step, min_mic_level, max_mic_level);
      if (level > new_level) {
        return level - new_level;
      }
--- a/modules/audio_processing/agc2/gain_applier.cc
+++ b/modules/audio_processing/agc2/gain_applier.cc
@ -28,7 +28,7 @@ void ClipSignal(DeinterleavedView<float> signal) {
  for (size_t k = 0; k < signal.num_channels(); ++k) {
    MonoView<float> channel_view = signal[k];
    for (auto& sample : channel_view) {
-      sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
+      sample = SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
    }
  }
 }
--- a/modules/audio_processing/agc2/input_volume_controller.cc
+++ b/modules/audio_processing/agc2/input_volume_controller.cc
@ -116,8 +116,8 @@ int GetSpeechLevelRmsErrorDb(float speech_level_dbfs,
  constexpr float kMaxSpeechLevelDbfs = 30.0f;
  RTC_DCHECK_GE(speech_level_dbfs, kMinSpeechLevelDbfs);
  RTC_DCHECK_LE(speech_level_dbfs, kMaxSpeechLevelDbfs);
-  speech_level_dbfs = rtc::SafeClamp<float>(
-      speech_level_dbfs, kMinSpeechLevelDbfs, kMaxSpeechLevelDbfs);
+  speech_level_dbfs = SafeClamp<float>(speech_level_dbfs, kMinSpeechLevelDbfs,
+                                       kMaxSpeechLevelDbfs);

  int rms_error_db = 0;
  if (speech_level_dbfs > target_range_max_dbfs) {
@ -343,7 +343,7 @@ void MonoInputVolumeController::UpdateInputVolume(int rms_error_db) {
  // Prevent too large microphone input volume changes by clamping the RMS
  // error.
  rms_error_db =
-      rtc::SafeClamp(rms_error_db, -KMaxAbsRmsErrorDbfs, KMaxAbsRmsErrorDbfs);
+      SafeClamp(rms_error_db, -KMaxAbsRmsErrorDbfs, KMaxAbsRmsErrorDbfs);
  if (rms_error_db == 0) {
    return;
  }
--- a/modules/audio_processing/agc2/input_volume_controller_unittest.cc
+++ b/modules/audio_processing/agc2/input_volume_controller_unittest.cc
@ -173,8 +173,8 @@ class SpeechSamplesReader {
      // Apply gain and copy samples into `audio_buffer_`.
      std::transform(buffer_.begin(), buffer_.end(),
                     audio_buffer_.channels()[0], [gain](int16_t v) -> float {
-                       return rtc::SafeClamp(static_cast<float>(v) * gain,
-                                             kMinSample, kMaxSample);
+                       return SafeClamp(static_cast<float>(v) * gain,
+                                        kMinSample, kMaxSample);
                     });
      controller.AnalyzeInputAudio(applied_input_volume, audio_buffer_);
      const auto recommended_input_volume = controller.RecommendInputVolume(
--- a/modules/audio_processing/agc2/limiter.cc
+++ b/modules/audio_processing/agc2/limiter.cc
@ -79,8 +79,8 @@ void ScaleSamples(MonoView<const float> per_sample_scaling_factors,
  for (size_t i = 0; i < signal.num_channels(); ++i) {
    MonoView<float> channel = signal[i];
    for (int j = 0; j < samples_per_channel; ++j) {
-      channel[j] = rtc::SafeClamp(channel[j] * per_sample_scaling_factors[j],
-                                  kMinFloatS16Value, kMaxFloatS16Value);
+      channel[j] = SafeClamp(channel[j] * per_sample_scaling_factors[j],
+                             kMinFloatS16Value, kMaxFloatS16Value);
    }
  }
 }
--- a/modules/audio_processing/agc2/saturation_protector.cc
+++ b/modules/audio_processing/agc2/saturation_protector.cc
@ -88,7 +88,7 @@ void UpdateSaturationProtectorState(float peak_dbfs,
  }

  state.headroom_db =
-      rtc::SafeClamp<float>(state.headroom_db, kMinMarginDb, kMaxMarginDb);
+      SafeClamp<float>(state.headroom_db, kMinMarginDb, kMaxMarginDb);
 }

 // Saturation protector which recommends a headroom based on the recent peaks.
--- a/modules/audio_processing/agc2/speech_level_estimator.cc
+++ b/modules/audio_processing/agc2/speech_level_estimator.cc
@ -20,7 +20,7 @@ namespace webrtc {
 namespace {

 float ClampLevelEstimateDbfs(float level_estimate_dbfs) {
-  return rtc::SafeClamp<float>(level_estimate_dbfs, -90.0f, 30.0f);
+  return SafeClamp<float>(level_estimate_dbfs, -90.0f, 30.0f);
 }

 // Returns the initial speech level estimate needed to apply the initial gain.
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@ -670,13 +670,13 @@ void ApmTest::ProcessDelayVerificationTest(int delay_ms,
  // Calculate expected delay estimate and acceptable regions. Further,
  // limit them w.r.t. AEC delay estimation support.
  const size_t samples_per_ms =
-      rtc::SafeMin<size_t>(16u, frame_.samples_per_channel() / 10);
+      SafeMin<size_t>(16u, frame_.samples_per_channel() / 10);
  const int expected_median =
-      rtc::SafeClamp<int>(delay_ms - system_delay_ms, delay_min, delay_max);
-  const int expected_median_high = rtc::SafeClamp<int>(
+      SafeClamp<int>(delay_ms - system_delay_ms, delay_min, delay_max);
+  const int expected_median_high = SafeClamp<int>(
      expected_median + rtc::dchecked_cast<int>(96 / samples_per_ms), delay_min,
      delay_max);
-  const int expected_median_low = rtc::SafeClamp<int>(
+  const int expected_median_low = SafeClamp<int>(
      expected_median - rtc::dchecked_cast<int>(96 / samples_per_ms), delay_min,
      delay_max);
  // Verify delay metrics.
--- a/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.cc
+++ b/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.cc
@ -84,7 +84,7 @@ void AudioSamplesScaler::Process(AudioBuffer& audio_buffer) {
    for (float& sample : channel_view) {
      constexpr float kMinFloatS16Value = -32768.f;
      constexpr float kMaxFloatS16Value = 32767.f;
-      sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
+      sample = SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
    }
  }
 }
--- a/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.cc
+++ b/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.cc
@ -80,7 +80,7 @@ void CaptureLevelsAdjuster::SetAnalogMicGainLevel(int level) {
  RTC_DCHECK_GE(level, kMinAnalogMicGainLevel);
  RTC_DCHECK_LE(level, kMaxAnalogMicGainLevel);
  int clamped_level =
-      rtc::SafeClamp(level, kMinAnalogMicGainLevel, kMaxAnalogMicGainLevel);
+      SafeClamp(level, kMinAnalogMicGainLevel, kMaxAnalogMicGainLevel);

  emulated_analog_mic_gain_level_ = clamped_level;
  UpdatePreAdjustmentGain();
--- a/modules/audio_processing/test/fake_recording_device.cc
+++ b/modules/audio_processing/test/fake_recording_device.cc
@ -87,8 +87,8 @@ class FakeRecordingDeviceLinear final : public FakeRecordingDeviceWorker {
    for (size_t c = 0; c < buffer->num_channels(); ++c) {
      for (size_t i = 0; i < buffer->num_frames(); ++i) {
        buffer->channels()[c][i] =
-            rtc::SafeClamp(buffer->channels()[c][i] * mic_level_ / divisor,
-                           kFloatSampleMin, kFloatSampleMax);
+            SafeClamp(buffer->channels()[c][i] * mic_level_ / divisor,
+                      kFloatSampleMin, kFloatSampleMax);
      }
    }
  }
@ -125,8 +125,8 @@ class FakeRecordingDeviceAgc final : public FakeRecordingDeviceWorker {
    for (size_t c = 0; c < buffer->num_channels(); ++c) {
      for (size_t i = 0; i < buffer->num_frames(); ++i) {
        buffer->channels()[c][i] =
-            rtc::SafeClamp(buffer->channels()[c][i] * scaling_factor,
-                           kFloatSampleMin, kFloatSampleMax);
+            SafeClamp(buffer->channels()[c][i] * scaling_factor,
+                      kFloatSampleMin, kFloatSampleMax);
      }
    }
  }
--- a/modules/congestion_controller/goog_cc/trendline_estimator.cc
+++ b/modules/congestion_controller/goog_cc/trendline_estimator.cc
@ -325,7 +325,7 @@ void TrendlineEstimator::UpdateThreshold(double modified_trend,
  const int64_t kMaxTimeDeltaMs = 100;
  int64_t time_delta_ms = std::min(now_ms - last_update_ms_, kMaxTimeDeltaMs);
  threshold_ += k * (fabs(modified_trend) - threshold_) * time_delta_ms;
-  threshold_ = rtc::SafeClamp(threshold_, 6.f, 600.f);
+  threshold_ = SafeClamp(threshold_, 6.f, 600.f);
  last_update_ms_ = now_ms;
 }

--- a/modules/remote_bitrate_estimator/overuse_detector.cc
+++ b/modules/remote_bitrate_estimator/overuse_detector.cc
@ -93,7 +93,7 @@ void OveruseDetector::UpdateThreshold(double modified_offset, int64_t now_ms) {
  const int64_t kMaxTimeDeltaMs = 100;
  int64_t time_delta_ms = std::min(now_ms - last_update_ms_, kMaxTimeDeltaMs);
  threshold_ += k * (fabs(modified_offset) - threshold_) * time_delta_ms;
-  threshold_ = rtc::SafeClamp(threshold_, 6.f, 600.f);
+  threshold_ = SafeClamp(threshold_, 6.f, 600.f);
  last_update_ms_ = now_ms;
 }

--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@ -420,15 +420,15 @@ std::vector<std::unique_ptr<RtpPacketToSend>> RTPSender::GeneratePadding(
      max_packet_size_ - max_padding_fec_packet_header_;
  if (audio_configured_) {
    // Allow smaller padding packets for audio.
-    padding_bytes_in_packet = rtc::SafeClamp<size_t>(
-        bytes_left, kMinAudioPaddingLength,
-        rtc::SafeMin(max_payload_size, kMaxPaddingLength));
+    padding_bytes_in_packet =
+        SafeClamp<size_t>(bytes_left, kMinAudioPaddingLength,
+                          SafeMin(max_payload_size, kMaxPaddingLength));
  } else {
    // Always send full padding packets. This is accounted for by the
    // RtpPacketSender, which will make sure we don't send too much padding even
    // if a single packet is larger than requested.
    // We do this to avoid frequently sending small packets on higher bitrates.
-    padding_bytes_in_packet = rtc::SafeMin(max_payload_size, kMaxPaddingLength);
+    padding_bytes_in_packet = SafeMin(max_payload_size, kMaxPaddingLength);
  }

  while (bytes_left > 0) {
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@ -145,6 +145,7 @@ rtc_library("basic_port_allocator") {

 rtc_source_set("candidate_pair_interface") {
  sources = [ "base/candidate_pair_interface.h" ]
+  deps = [ "../api:candidate" ]
 }

 rtc_library("connection") {
--- a/p2p/base/candidate_pair_interface.h
+++ b/p2p/base/candidate_pair_interface.h
@ -11,9 +11,9 @@
 #ifndef P2P_BASE_CANDIDATE_PAIR_INTERFACE_H_
 #define P2P_BASE_CANDIDATE_PAIR_INTERFACE_H_

-namespace cricket {
+#include "api/candidate.h"

-class Candidate;
+namespace cricket {

 class CandidatePairInterface {
 public:
--- a/p2p/base/connection.cc
+++ b/p2p/base/connection.cc
@ -983,7 +983,7 @@ void Connection::UpdateState(int64_t now) {
    return;

  // Computes our estimate of the RTT given the current estimate.
-  int rtt = rtc::SafeClamp(2 * rtt_, MINIMUM_RTT, MAXIMUM_RTT);
+  int rtt = webrtc::SafeClamp(2 * rtt_, MINIMUM_RTT, MAXIMUM_RTT);

  if (RTC_LOG_CHECK_LEVEL(LS_VERBOSE)) {
    std::string pings;
--- a/p2p/base/pseudo_tcp.cc
+++ b/p2p/base/pseudo_tcp.cc
@ -713,8 +713,8 @@ bool PseudoTcp::process(Segment& seg) {
          m_rx_rttvar = (3 * m_rx_rttvar + abs_err) / 4;
          m_rx_srtt = (7 * m_rx_srtt + rtt) / 8;
        }
-        m_rx_rto = rtc::SafeClamp(m_rx_srtt + rtc::SafeMax(1, 4 * m_rx_rttvar),
-                                  MIN_RTO, MAX_RTO);
+        m_rx_rto = webrtc::SafeClamp(
+            m_rx_srtt + webrtc::SafeMax(1, 4 * m_rx_rttvar), MIN_RTO, MAX_RTO);
 #if _DEBUGMSG >= _DBG_VERBOSE
        RTC_LOG(LS_INFO) << "rtt: " << rtt << "  srtt: " << m_rx_srtt
                         << "  rto: " << m_rx_rto;
--- a/pc/jitter_buffer_delay.cc
+++ b/pc/jitter_buffer_delay.cc
@ -29,10 +29,9 @@ void JitterBufferDelay::Set(std::optional<double> delay_seconds) {

 int JitterBufferDelay::GetMs() const {
  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
-  return rtc::SafeClamp(
-      rtc::saturated_cast<int>(cached_delay_seconds_.value_or(kDefaultDelay) *
-                               1000),
-      0, kMaximumDelayMs);
+  return SafeClamp(rtc::saturated_cast<int>(
+                       cached_delay_seconds_.value_or(kDefaultDelay) * 1000),
+                   0, kMaximumDelayMs);
 }

 }  // namespace webrtc
--- a/rtc_base/numerics/safe_minmax.h
+++ b/rtc_base/numerics/safe_minmax.h
@ -84,7 +84,7 @@
 #include "rtc_base/numerics/safe_compare.h"
 #include "rtc_base/type_traits.h"

-namespace rtc {
+namespace webrtc {

 namespace safe_minmax_impl {

@ -331,6 +331,14 @@ R2 SafeClamp(T x, L min, H max) {
                          : static_cast<R2>(x);
 }

+}  //  namespace webrtc
+
+// Re-export symbols from the webrtc namespace for backwards compatibility.
+// TODO(bugs.webrtc.org/4222596): Remove once all references are updated.
+namespace rtc {
+using ::webrtc::SafeClamp;
+using ::webrtc::SafeMax;
+using ::webrtc::SafeMin;
 }  // namespace rtc

 #endif  // RTC_BASE_NUMERICS_SAFE_MINMAX_H_
--- a/rtc_base/numerics/safe_minmax_unittest.cc
+++ b/rtc_base/numerics/safe_minmax_unittest.cc
@ -17,7 +17,7 @@

 #include "test/gtest.h"

-namespace rtc {
+namespace webrtc {

 namespace {

@ -344,4 +344,4 @@ uint32_t TestClampSafe(uint32_t x, uint32_t a, uint32_t b) {
  return SafeClamp(x, a, b);
 }

-}  // namespace rtc
+}  // namespace webrtc
--- a/rtc_base/strings/string_builder.cc
+++ b/rtc_base/strings/string_builder.cc
@ -34,8 +34,7 @@ SimpleStringBuilder& SimpleStringBuilder::operator<<(char ch) {
 SimpleStringBuilder& SimpleStringBuilder::operator<<(absl::string_view str) {
  RTC_DCHECK_LT(size_ + str.length(), buffer_.size())
      << "Buffer size was insufficient";
-  const size_t chars_added =
-      rtc::SafeMin(str.length(), buffer_.size() - size_ - 1);
+  const size_t chars_added = SafeMin(str.length(), buffer_.size() - size_ - 1);
  memcpy(&buffer_[size_], str.data(), chars_added);
  size_ += chars_added;
  buffer_[size_] = '\0';
@ -97,7 +96,7 @@ SimpleStringBuilder& SimpleStringBuilder::AppendFormat(const char* fmt, ...) {
  const int len =
      std::vsnprintf(&buffer_[size_], buffer_.size() - size_, fmt, args);
  if (len >= 0) {
-    const size_t chars_added = rtc::SafeMin(len, buffer_.size() - 1 - size_);
+    const size_t chars_added = SafeMin(len, buffer_.size() - 1 - size_);
    size_ += chars_added;
    RTC_DCHECK_EQ(len, chars_added) << "Buffer size was insufficient";
  } else {
--- a/sdk/android/src/jni/audio_device/audio_device_module.cc
+++ b/sdk/android/src/jni/audio_device/audio_device_module.cc
@ -298,8 +298,6 @@ class AndroidAudioDeviceModule : public AudioDeviceModule {
    RTC_DLOG(LS_INFO) << __FUNCTION__;
    if (!initialized_)
      return -1;
-    if (!Recording())
-      return 0;
    audio_device_buffer_->StopRecording();
    int32_t result = input_->StopRecording();
    RTC_DLOG(LS_INFO) << "output: " << result;
--- a/test/network/cross_traffic.cc
+++ b/test/network/cross_traffic.cc
@ -56,7 +56,7 @@ void RandomWalkCrossTraffic::Process(Timestamp at_time) {
  if (at_time - last_update_time_ >= config_.update_interval) {
    intensity_ += random_.Gaussian(config_.bias, config_.variance) *
                  sqrt((at_time - last_update_time_).seconds<double>());
-    intensity_ = rtc::SafeClamp(intensity_, 0.0, 1.0);
+    intensity_ = SafeClamp(intensity_, 0.0, 1.0);
    last_update_time_ = at_time;
  }
  pending_size_ += TrafficRate() * delta;
--- a/video/corruption_detection/halton_frame_sampler.cc
+++ b/video/corruption_detection/halton_frame_sampler.cc
@ -137,7 +137,7 @@ double GetFilteredElement(int width,
  }

  // Take the rounding errors into consideration.
-  return rtc::SafeClamp(element_sum / total_weight, 0.0, 255.0);
+  return SafeClamp(element_sum / total_weight, 0.0, 255.0);
 }

 std::vector<FilteredSample> GetSampleValuesForFrame(
--- a/webrtc.gni
+++ b/webrtc.gni
@ -170,18 +170,24 @@ declare_args() {
  rtc_build_with_neon =
      (current_cpu == "arm" && arm_use_neon) || current_cpu == "arm64"

-  # Enable this to build OpenH264 encoder/FFmpeg decoder. This is supported on
-  # all platforms except Android and iOS. Because FFmpeg can be built
-  # with/without H.264 support, `ffmpeg_branding` has to separately be set to a
-  # value that includes H.264, for example "Chrome". If FFmpeg is built without
-  # H.264, compilation succeeds but `H264DecoderImpl` fails to initialize.
+  # Enable this to build OpenH264 encoder/FFmpeg decoder. When building WebRTC
+  # as part of Chromium, this is delegated to `media_use_openh264`. When
+  # building WebRTC as a standalone library, this is supported on all platforms
+  # except Android and iOS. Because FFmpeg can be built with/without H.264
+  # support, `ffmpeg_branding` has to separately be set to a value that
+  # includes H.264, for example "Chrome". If FFmpeg is built without H.264,
+  # compilation succeeds but `H264DecoderImpl` fails to initialize.
  # CHECK THE OPENH264, FFMPEG AND H.264 LICENSES/PATENTS BEFORE BUILDING.
  # http://www.openh264.org, https://www.ffmpeg.org/
  #
  # Enabling H264 when building with MSVC is currently not supported, see
  # bugs.webrtc.org/9213#c13 for more info.
-  rtc_use_h264 =
-      proprietary_codecs && !is_android && !is_ios && !(is_win && !is_clang)
+  if (build_with_chromium) {
+    rtc_use_h264 = media_use_openh264
+  } else {
+    rtc_use_h264 =
+        proprietary_codecs && !is_android && !is_ios && !(is_win && !is_clang)
+  }

  # Use system OpenH264
  rtc_system_openh264 = false
Author	SHA1	Message	Date
Jeremy Leconte	8f573d6960	Update iOS version from 17 to 18. Change-Id: Id4b782f6f10dca3e5cc7f34c2f6620253b3f24d0 Bug: None Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378042 Reviewed-by: Christoffer Dewerin <jansson@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43949}	2025-02-21 01:42:29 -08:00
chromium-webrtc-autoroll	a756a7ad15	Roll chromium_revision c50f9710da..40885069c1 (1422420:1422971) Change log: `c50f9710da..40885069c1` Full diff: `c50f9710da..40885069c1` Changed dependencies * src/base: `9e81d0aac9..a62135f7fa` * src/build: `e42ba23e14..530d40d7d3` * src/buildtools: `d1ed719a33..99bd087289` * src/ios: `f497e652b6..f8fdd3967c` * src/testing: `75065cfcbb..0d1f0f2d5d` * src/third_party: `f004cc0a39..adf98b9a1e` * src/third_party/androidx/cipd: R1wNTeO5kH2bzp1NW4g87DLWtiDXaImAQJplHSR-108C..9i2Jdh3nRH-lCfGUTvpGecSiz9lyJ3mGDYTNz7N04csC * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/294ab9730c..0c59e9e55b * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/30345af7ee..09f963b5f4 * src/third_party/depot_tools: `ab387d1ddc..2259aa1766` * src/third_party/fuzztest/src: `1d61720657..409e5199f6` * src/third_party/libc++/src: `e36be6784f..1a88911caa` * src/third_party/llvm-libc/src: `99782c3117..e722dac269` * src/third_party/perfetto: `1e97ffd153..072b48f0e0` * src/third_party/r8/cipd: QDwsH_fhjzc4w9ejy5IKJrAl9g6ct71e1UJiFwGoqYMC..2ChXGrcBJ9WNhFhS069j3L6VMp0jw5xyyPqKhlOqj_sC * src/third_party/rust-toolchain_version: Linux_x64/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Mac/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Mac_arm64/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz,Win/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz..Linux_x64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac_arm64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Win/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz * src/tools: `367e383074..a4cad1e12a` DEPS diff: `c50f9710da..40885069c1`/DEPS No update to Clang. BUG=None Change-Id: I764ffb6cc53a7d0c2a498d185c79bdb12cda5d90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378300 Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#43948}	2025-02-21 00:37:55 -08:00
Jeremy Leconte	e90ab59b7c	Revert "Move resources to resources/BUILD.gn." This reverts commit 7dea26d8bb0fbb2f6fe25e74d2baac9293e413a8. Reason for revert: breaks downstream Original change's description: > Move resources to resources/BUILD.gn. > > iOS bundle all resources in the same folder and some conflicts can arise from that. > Having all resources in the same file makes it easier to reason about it. > > Change-Id: I37f420dfbd265ec644804e9d4c96515c83d2a992 > Bug: b/397385850 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377821 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Jeremy Leconte <jleconte@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#43944} Bug: b/397385850 Change-Id: I80788590498fc24709c95a6a9580fdad65860f8c No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378280 Commit-Queue: Jeremy Leconte <jleconte@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43947}	2025-02-21 00:01:13 -08:00
Jonas Oreland	358c94f5dd	AudioState extensions This patch modifies AudioState to always call InitRecording before StartRecording(). This makes it possible to do SetRecording(false) + SetRecording(true), which before this patch would not actually work if there was sending streams. The only way was to add/remove streams...via SDP operations, puh :(. Bonus: We also needed to modifu AndroidAudioDeviceModule (which is a thin wrapper) so that StopRecording() will call AudioInput->StopRecording() even when recording is not enabled. BUG=b/397376626 Change-Id: I954b5caab11225b544c3e6a78c5dde357d4eedb5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378140 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Auto-Submit: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43946}	2025-02-20 23:50:12 -08:00
webrtc-version-updater	21b5db48da	Update WebRTC code version (2025-02-21T04:05:42). Bug: None Change-Id: I5d7a91092f6b6d059a02244e198f4381f05e9b5e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378260 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#43945}	2025-02-20 21:44:16 -08:00
Jeremy Leconte	7dea26d8bb	Move resources to resources/BUILD.gn. iOS bundle all resources in the same folder and some conflicts can arise from that. Having all resources in the same file makes it easier to reason about it. Change-Id: I37f420dfbd265ec644804e9d4c96515c83d2a992 Bug: b/397385850 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377821 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43944}	2025-02-20 08:48:47 -08:00
Jeremy Leconte	cea5e723b5	Re-enable iOS simulator bots. Change-Id: I2e05dd6f40467ee94072a0daac94142d50ff8032 Bug: b/396118151 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378020 Commit-Queue: Jeremy Leconte <jleconte@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43943}	2025-02-20 07:48:26 -08:00
Evan Shrubsole	d9dd939d66	Move safe_minmax.h to webrtc namespace Bug: webrtc:42232595 Change-Id: Ia3d96dfe1b1c25b6cc21bbd99d24ded7461924cd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378061 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43942}	2025-02-20 05:12:52 -08:00
Philipp Hancke	ad82b6e45b	IWYU candidate_pair_interface and remove the forward declaration BUG=webrtc:42226242 Change-Id: I6bff2ef5048e641c834f9e03620ce038f6513065 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376920 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#43941}	2025-02-20 05:11:14 -08:00
Mosa Morosev	b300a9cc05	OpenH264 library enabled in Android Chromium build H264 codec components are included in Chromium build on Android when OpenH264 library is enabled in Chromium media using media_use_openh264 build flag. Bug: chromium:40519162 Change-Id: I4fcc341a7d208b399a294b1f86774860f70f83b6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43940}	2025-02-20 03:51:15 -08:00