sendbird · sf-jed-kyung · Jul 31, 2025 · Aug 4, 2025 · Aug 4, 2025 · Aug 4, 2025
diff --git a/.clang-tidy b/.clang-tidy
@@ -0,0 +1,17 @@
+---
+---
+  Checks:          '-*,
+                    bugprone-argument-comment
+                    llvm-namespace-comment,
+                    modernize-use-designated-initializers,
+                    modernize-use-override,
+                    modernize-use-std-numbers,
+                    readability-container-size-empty,
+                    readability-static-definition-in-anonymous-namespace,
+                    readability-redundant-smartptr-get,
+                    readability-redundant-declaration'
+                    # readability-static-accessed-through-instance is generally useful but produces too many false positives.
+  CheckOptions:
+    # The default threshold gives false positives in test data
+    modernize-use-std-numbers.DiffThreshold: 0.00001
+...
diff --git a/.gn b/.gn
@@ -55,7 +55,7 @@ default_args = {
   ios_deployment_target = "14.0"
 
   # The SDK API level, in contrast, is set by build/android/AndroidManifest.xml.
-  android_ndk_api_level = 21
+  android_ndk_api_level = 23
 
   # WebRTC does not provide the gflags dependency. Because libyuv uses it only
   # for unittests, it can be disabled (see third_party/libyuv/BUILD.gn)
@@ -66,7 +66,7 @@ default_args = {
   gtest_enable_absl_printers = true
 
   # Differently from Chromium, WebRTC still support SDK 21.
-  default_min_sdk_version = 21
+  default_min_sdk_version = 23
 
   # Prevent jsoncpp to pass -Wno-deprecated-declarations to users
   jsoncpp_no_deprecated_declarations = false

diff --git a/.vpython3 b/.vpython3
@@ -80,7 +80,7 @@ wheel: <
 >
 wheel: <
   name: "infra/python/wheels/protobuf-py3"
-  version: "version:6.31.0"
+  version: "version:6.32.0"
 >
 wheel: <
   name: "infra/python/wheels/requests-py3"

diff --git a/BUILD.gn b/BUILD.gn
@@ -372,6 +372,10 @@ config("common_config") {
     defines += [ "WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE" ]
   }
 
+  if (rtc_video_psnr) {
+    defines += [ "WEBRTC_ENCODER_PSNR_STATS" ]
+  }
+
   if (is_clang) {
     cflags += [
       "-Wshadow",
@@ -660,6 +664,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "rtc_base:rtc_numerics_unittests",
       "rtc_base:rtc_operations_chain_unittests",
       "rtc_base:rtc_task_queue_unittests",
+      "rtc_base:sigslot_trampoline_unittest",
       "rtc_base:sigslot_unittest",
       "rtc_base:task_queue_stdlib_unittest",
       "rtc_base:untyped_function_unittest",

diff --git a/DEPS b/DEPS
diff --git a/api/BUILD.gn b/api/BUILD.gn
@@ -70,14 +70,20 @@ rtc_library("enable_media_with_defaults") {
   ]
 }
 
-rtc_source_set("create_modular_peer_connection_factory") {
+rtc_library("create_modular_peer_connection_factory") {
   visibility = [ "*" ]
   allow_poison = [ "environment_construction" ]
-  sources = [ "create_modular_peer_connection_factory.h" ]
+  sources = [
+    "create_modular_peer_connection_factory.cc",
+    "create_modular_peer_connection_factory.h",
+  ]
   deps = [
     ":libjingle_peerconnection_api",
     ":scoped_refptr",
+    ":sequence_checker",
     "../pc:peer_connection_factory",
+    "../pc:peer_connection_factory_proxy",
+    "../rtc_base:threading",
     "../rtc_base/system:rtc_export",
     "../stats:rtc_stats",
   ]
@@ -233,6 +239,7 @@ rtc_source_set("ice_transport_interface") {
     ":ref_count",
     ":rtc_error",
     ":scoped_refptr",
+    "environment",
     "rtc_event_log",
   ]
 }
@@ -289,17 +296,6 @@ rtc_library("rtp_sender_interface") {
   ]
 }
 
-# TODO: bugs.webrtc.org/42220069 - Remove when downstream has been updated
-# to depend on `create_modular_peer_connection_factory` instead.
-rtc_source_set("create_modular_peer_connection_factory_internal") {
-  visibility = [ ":libjingle_peerconnection_api" ]
-  sources = [ "create_modular_peer_connection_factory_internal.h" ]
-  deps = [
-    ":scoped_refptr",
-    "../rtc_base/system:rtc_export",
-  ]
-}
-
 rtc_library("libjingle_peerconnection_api") {
   visibility = [ "*" ]
   cflags = []
@@ -340,7 +336,6 @@ rtc_library("libjingle_peerconnection_api") {
     ":async_dns_resolver",
     ":audio_options_api",
     ":candidate",
-    ":create_modular_peer_connection_factory_internal",
     ":data_channel_event_observer_interface",
     ":dtls_transport_interface",
     ":fec_controller_api",
@@ -1653,6 +1648,9 @@ if (rtc_include_tests) {
       "video:rtp_video_frame_assembler_unittests",
       "video:video_frame",
       "video:video_frame_metadata_unittest",
+      "video/corruption_detection:frame_instrumentation_data_reader_unittest",
+      "video/corruption_detection:frame_instrumentation_data_unittest",
+      "video/corruption_detection:frame_instrumentation_evaluation_unittest",
       "//third_party/abseil-cpp/absl/functional:any_invocable",
       "//third_party/abseil-cpp/absl/strings",
       "//third_party/abseil-cpp/absl/strings:string_view",

diff --git a/api/async_dns_resolver.h b/api/async_dns_resolver.h
@@ -23,20 +23,18 @@ namespace webrtc {
 // The AsyncDnsResolverInterface class encapsulates a single name query.
 //
 // Usage:
-//   std::unique_ptr<AsyncDnsResolverInterface> resolver =
-//        factory->Create(address-to-be-resolved, [r = resolver.get()]() {
-//     if (r->result.GetResolvedAddress(AF_INET, &addr) {
-//       // success
-//     } else {
-//       // failure
-//       error = r->result().GetError();
-//     }
-//     // Release resolver.
-//     resolver_list.erase(std::remove_if(resolver_list.begin(),
-//     resolver_list.end(),
-//                         [](refptr) { refptr.get() == r; });
-//   });
-//   resolver_list.push_back(std::move(resolver));
+// std::unique_ptr<AsyncDnsResolverInterface> resolver =
+//      factory->Create(address-to-be-resolved, [r = resolver.get()]() {
+//   if (r->result.GetResolvedAddress(AF_INET, &addr) {
+//     // success
+//   } else {
+//     // failure
+//     error = r->result().GetError();
+//   }
+//   // Release resolver.
+//   std::erase_if(resolver_list, [](refptr) { refptr.get() == r; });
+// });
+// resolver_list.push_back(std::move(resolver));
 
 class AsyncDnsResolverResult {
  public:

diff --git a/api/audio_codecs/audio_decoder.cc b/api/audio_codecs/audio_decoder.cc
@@ -48,7 +48,8 @@ class OldStyleEncodedFrame final : public AudioDecoder::EncodedAudioFrame {
         decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
     return ret < 0 ? std::nullopt
                    : std::optional<DecodeResult>(
-                         {static_cast<size_t>(ret), speech_type});
+                         {.num_decoded_samples = static_cast<size_t>(ret),
+                          .speech_type = speech_type});
   }
 
  private:
@@ -174,5 +175,4 @@ AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) {
   }
 }
 
-constexpr int AudioDecoder::kMaxNumberOfChannels;
 }  // namespace webrtc
diff --git a/api/audio_codecs/audio_encoder.cc b/api/audio_codecs/audio_encoder.cc
@@ -14,8 +14,8 @@
 #include <cstdint>
 #include <memory>
 #include <optional>
-#include <string>
 
+#include "absl/strings/string_view.h"
 #include "api/array_view.h"
 #include "api/call/bitrate_allocation.h"
 #include "rtc_base/buffer.h"
@@ -82,9 +82,7 @@ AudioEncoder::ReclaimContainedEncoders() {
   return nullptr;
 }
 
-bool AudioEncoder::EnableAudioNetworkAdaptor(
-    const std::string& /* config_string */,
-    RtcEventLog* /* event_log */) {
+bool AudioEncoder::EnableAudioNetworkAdaptor(absl::string_view /*config*/) {
   return false;
 }
 
@@ -122,5 +120,4 @@ ANAStats AudioEncoder::GetANAStats() const {
   return ANAStats();
 }
 
-constexpr int AudioEncoder::kMaxNumberOfChannels;
 }  // namespace webrtc
diff --git a/api/audio_codecs/audio_encoder.h b/api/audio_codecs/audio_encoder.h
@@ -16,11 +16,11 @@
 
 #include <memory>
 #include <optional>
-#include <string>
 #include <utility>
 #include <vector>
 
 #include "absl/base/attributes.h"
+#include "absl/strings/string_view.h"
 #include "api/array_view.h"
 #include "api/audio/audio_view.h"
 #include "api/call/bitrate_allocation.h"
@@ -30,8 +30,6 @@
 
 namespace webrtc {
 
-class RtcEventLog;
-
 // Statistics related to Audio Network Adaptation.
 struct ANAStats {
   ANAStats();
@@ -201,8 +199,7 @@ class AudioEncoder {
   virtual ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders();
 
   // Enables audio network adaptor. Returns true if successful.
-  virtual bool EnableAudioNetworkAdaptor(const std::string& config_string,
-                                         RtcEventLog* event_log);
+  virtual bool EnableAudioNetworkAdaptor(absl::string_view config);
 
   // Disables audio network adaptor.
   virtual void DisableAudioNetworkAdaptor();

diff --git a/api/audio_codecs/opus/audio_encoder_opus_config.cc b/api/audio_codecs/opus/audio_encoder_opus_config.cc
@@ -27,10 +27,6 @@ constexpr int kDefaultLowRateComplexity =
 
 }  // namespace
 
-constexpr int AudioEncoderOpusConfig::kDefaultFrameSizeMs;
-constexpr int AudioEncoderOpusConfig::kMinBitrateBps;
-constexpr int AudioEncoderOpusConfig::kMaxBitrateBps;
-
 AudioEncoderOpusConfig::AudioEncoderOpusConfig()
     : frame_size_ms(kDefaultFrameSizeMs),
       sample_rate_hz(48000),

diff --git a/api/call/bitrate_allocation.h b/api/call/bitrate_allocation.h
@@ -27,7 +27,7 @@ struct BitrateAllocationUpdate {
   double packet_loss_ratio = 0;
   // Predicted round trip time.
   TimeDelta round_trip_time = TimeDelta::PlusInfinity();
-  // `bwe_period` is deprecated, use `stable_target_bitrate` allocation instead.
+  // `bwe_period` is deprecated.
   TimeDelta bwe_period = TimeDelta::PlusInfinity();
   // Congestion window pushback bitrate reduction fraction. Used in
   // VideoStreamEncoder to reduce the bitrate by the given fraction

diff --git a/api/candidate.h b/api/candidate.h
@@ -192,8 +192,12 @@ class RTC_EXPORT Candidate {
   void set_tcptype(absl::string_view tcptype) { Assign(tcptype_, tcptype); }
 
   // The name of the transport channel of this candidate.
-  // TODO(phoglund): remove.
-  const std::string& transport_name() const { return transport_name_; }
+  // TODO(bugs.webrtc.org/42233526): remove.
+  [[deprecated("Use IceCandidate::sdp_mid")]]
+  const std::string& transport_name() const {
+    return transport_name_;
+  }
+  [[deprecated("Use the IceCandidate type for sdp_mid")]]
   void set_transport_name(absl::string_view transport_name) {
     Assign(transport_name_, transport_name);
   }

diff --git a/api/create_modular_peer_connection_factory.cc b/api/create_modular_peer_connection_factory.cc
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2025 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/create_modular_peer_connection_factory.h"
+
+#include <utility>
+
+#include "api/peer_connection_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "pc/peer_connection_factory.h"
+#include "pc/peer_connection_factory_proxy.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+
+scoped_refptr<PeerConnectionFactoryInterface>
+CreateModularPeerConnectionFactory(
+    PeerConnectionFactoryDependencies dependencies) {
+  // The PeerConnectionFactory must be created on the signaling thread.
+  if (dependencies.signaling_thread &&
+      !dependencies.signaling_thread->IsCurrent()) {
+    return dependencies.signaling_thread->BlockingCall([&dependencies] {
+      return CreateModularPeerConnectionFactory(std::move(dependencies));
+    });
+  }
+
+  auto pc_factory = PeerConnectionFactory::Create(std::move(dependencies));
+  if (!pc_factory) {
+    return nullptr;
+  }
+  // Verify that the invocation and the initialization ended up agreeing on the
+  // thread.
+  RTC_DCHECK_RUN_ON(pc_factory->signaling_thread());
+  return PeerConnectionFactoryProxy::Create(
+      pc_factory->signaling_thread(), pc_factory->worker_thread(), pc_factory);
+}
+
+}  // namespace webrtc
diff --git a/api/create_modular_peer_connection_factory_internal.h b/api/create_modular_peer_connection_factory_internal.h
diff --git a/api/crypto/BUILD.gn b/api/crypto/BUILD.gn
@@ -27,6 +27,7 @@ rtc_library("options") {
     "../../rtc_base:checks",
     "../../rtc_base:ssl_adapter",
     "../../rtc_base/system:rtc_export",
+    "//third_party/abseil-cpp/absl/algorithm:container",
   ]
 }