mozilla: comparison mozilla-silence-no-return-type.patch

equal deleted inserted replaced

-:c19c9e7820ef
+:d76083122710
 # HG changeset patch
-# Parent  602c790a8615e43dbfe8ce15a30d020e0fb4f5e7
+# Parent  ccd8f974707cba440cffeb0c66b5bcc0cda73c63
 diff --git a/Cargo.lock b/Cargo.lock
 --- a/Cargo.lock
 +++ b/Cargo.lock
-@@ -2298,18 +2298,16 @@ name = "glsl-to-cxx"
+@@ -2296,18 +2296,16 @@ name = "glsl-to-cxx"
 version = "0.1.0"
 dependencies = [
 "glsl",
 ]
 version = "0.1.0"
 dependencies = [
 diff --git a/Cargo.toml b/Cargo.toml
 --- a/Cargo.toml
 +++ b/Cargo.toml
-@@ -146,16 +146,17 @@ async-task = { git = "https://github.com
+@@ -143,16 +143,17 @@ async-task = { git = "https://github.com
 chardetng = { git = "https://github.com/hsivonen/chardetng", rev="3484d3e3ebdc8931493aa5df4d7ee9360a90e76b" }
 chardetng_c = { git = "https://github.com/hsivonen/chardetng_c", rev="ed8a4c6f900a90d4dbc1d64b856e61490a1c3570" }
 coremidi = { git = "https://github.com/chris-zen/coremidi.git", rev="fc68464b5445caf111e41f643a2e69ccce0b4f83" }
 fog = { path = "toolkit/components/glean/api" }
 libudev-sys = { path = "dom/webauthn/libudev-sys" }
 midir = { git = "https://github.com/mozilla/midir.git", rev = "e1b4dcb767f9e69afe95a860374aaa9635d81e3d" }
 minidump_writer_linux = { git = "https://github.com/rust-minidump/minidump-writer.git", rev = "75ada456c92a429704691a85e1cb42fef8cafc0d" }
 +glslopt = { path = "third_party/rust/glslopt/" }
 # application-services overrides to make updating them all simpler.
-interrupt-support = { git = "https://github.com/mozilla/application-services", rev = "2689788cecf24c385e6b7440e3aa1a89c511f14a" }
+interrupt-support = { git = "https://github.com/mozilla/application-services", rev = "fb1c78b13c27b5db1fd5458b8c2d8f433855dd61" }
-sql-support = { git = "https://github.com/mozilla/application-services", rev = "2689788cecf24c385e6b7440e3aa1a89c511f14a" }
+sql-support = { git = "https://github.com/mozilla/application-services", rev = "fb1c78b13c27b5db1fd5458b8c2d8f433855dd61" }
-sync15-traits = { git = "https://github.com/mozilla/application-services", rev = "2689788cecf24c385e6b7440e3aa1a89c511f14a" }
+sync15-traits = { git = "https://github.com/mozilla/application-services", rev = "fb1c78b13c27b5db1fd5458b8c2d8f433855dd61" }
-viaduct = { git = "https://github.com/mozilla/application-services", rev = "2689788cecf24c385e6b7440e3aa1a89c511f14a" }
+viaduct = { git = "https://github.com/mozilla/application-services", rev = "fb1c78b13c27b5db1fd5458b8c2d8f433855dd61" }
-webext-storage = { git = "https://github.com/mozilla/application-services", rev = "2689788cecf24c385e6b7440e3aa1a89c511f14a" }
+webext-storage = { git = "https://github.com/mozilla/application-services", rev = "fb1c78b13c27b5db1fd5458b8c2d8f433855dd61" }
 diff --git a/gfx/skia/skia/include/codec/SkEncodedOrigin.h b/gfx/skia/skia/include/codec/SkEncodedOrigin.h
 --- a/gfx/skia/skia/include/codec/SkEncodedOrigin.h
 +++ b/gfx/skia/skia/include/codec/SkEncodedOrigin.h
 @@ -36,12 +36,13 @@ static inline SkMatrix SkEncodedOriginTo
 Precision::constructFractionSignificant(
 const FractionPrecision &base,
 diff --git a/third_party/libwebrtc/api/adaptation/resource.cc b/third_party/libwebrtc/api/adaptation/resource.cc
 --- a/third_party/libwebrtc/api/adaptation/resource.cc
 +++ b/third_party/libwebrtc/api/adaptation/resource.cc
-@@ -4,25 +4,29 @@
+@@ -17,16 +17,17 @@ namespace webrtc {
-*  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/adaptation/resource.h"
-+#include "rtc_base/checks.h"
-namespace webrtc {
 const char* ResourceUsageStateToString(ResourceUsageState usage_state) {
 switch (usage_state) {
 case ResourceUsageState::kOveruse:
 return "kOveruse";
 case ResourceUsageState::kUnderuse:
 return "kUnderuse";
-+    default:
-+      RTC_NOTREACHED();
-+      return "";
 }
+RTC_CHECK_NOTREACHED();
++  return nullptr;
 }
 ResourceListener::~ResourceListener() {}
 Resource::Resource() {}
 Resource::~Resource() {}
 diff --git a/third_party/libwebrtc/api/rtp_parameters.cc b/third_party/libwebrtc/api/rtp_parameters.cc
 --- a/third_party/libwebrtc/api/rtp_parameters.cc
 +++ b/third_party/libwebrtc/api/rtp_parameters.cc
-@@ -24,16 +24,19 @@ const char* DegradationPreferenceToStrin
+@@ -27,16 +27,17 @@ const char* DegradationPreferenceToStrin
-case DegradationPreference::DISABLED:
-return "disabled";
 case DegradationPreference::MAINTAIN_FRAMERATE:
 return "maintain-framerate";
 case DegradationPreference::MAINTAIN_RESOLUTION:
 return "maintain-resolution";
 case DegradationPreference::BALANCED:
 return "balanced";
-+    default:
-+      RTC_NOTREACHED();
-+      return "";
 }
+RTC_CHECK_NOTREACHED();
++  return "";
 }
 const double kDefaultBitratePriority = 1.0;
 RtcpFeedback::RtcpFeedback() = default;
 RtcpFeedback::RtcpFeedback(RtcpFeedbackType type) : type(type) {}
 RtcpFeedback::RtcpFeedback(RtcpFeedbackType type,
+RtcpFeedbackMessageType message_type)
+diff --git a/third_party/libwebrtc/api/video/video_frame_buffer.cc b/third_party/libwebrtc/api/video/video_frame_buffer.cc
+--- a/third_party/libwebrtc/api/video/video_frame_buffer.cc
++++ b/third_party/libwebrtc/api/video/video_frame_buffer.cc
+@@ -87,16 +87,18 @@ const char* VideoFrameBufferTypeToString
+return "kI422";
+case VideoFrameBuffer::Type::kI010:
+return "kI010";
+case VideoFrameBuffer::Type::kNV12:
+return "kNV12";
+default:
+RTC_DCHECK_NOTREACHED();
+}
++  RTC_DCHECK_NOTREACHED();
++  return nullptr;
+}
+int I420BufferInterface::ChromaWidth() const {
+return (width() + 1) / 2;
+}
+int I420BufferInterface::ChromaHeight() const {
+return (height() + 1) / 2;
 diff --git a/third_party/libwebrtc/api/video_codecs/video_codec.cc b/third_party/libwebrtc/api/video_codecs/video_codec.cc
 --- a/third_party/libwebrtc/api/video_codecs/video_codec.cc
 +++ b/third_party/libwebrtc/api/video_codecs/video_codec.cc
-@@ -114,16 +114,19 @@ const char* CodecTypeToPayloadString(Vid
+@@ -117,16 +117,17 @@ const char* CodecTypeToPayloadString(Vid
-case kVideoCodecAV1:
-return kPayloadNameAv1;
 case kVideoCodecH264:
 return kPayloadNameH264;
 case kVideoCodecMultiplex:
 return kPayloadNameMultiplex;
 case kVideoCodecGeneric:
 return kPayloadNameGeneric;
-+    default:
-+      RTC_NOTREACHED();
-+      return "";
 }
+RTC_CHECK_NOTREACHED();
++  return "";
 }
 VideoCodecType PayloadStringToCodecType(const std::string& name) {
 if (absl::EqualsIgnoreCase(name, kPayloadNameVp8))
 return kVideoCodecVP8;
 if (absl::EqualsIgnoreCase(name, kPayloadNameVp9))
 return kVideoCodecVP9;
+if (absl::EqualsIgnoreCase(name, kPayloadNameAv1) ||
 diff --git a/third_party/libwebrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc b/third_party/libwebrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc
 --- a/third_party/libwebrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc
 +++ b/third_party/libwebrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc
-@@ -156,16 +156,19 @@ class VideoEncoderSoftwareFallbackWrappe
+@@ -158,16 +158,17 @@ class VideoEncoderSoftwareFallbackWrappe
-<< "Trying to access encoder in uninitialized fallback wrapper.";
+[[fallthrough]];
-// Return main encoder to preserve previous behavior.
-ABSL_FALLTHROUGH_INTENDED;
 case EncoderState::kMainEncoderUsed:
 return encoder_.get();
 case EncoderState::kFallbackDueToFailure:
 case EncoderState::kForcedFallback:
 return fallback_encoder_.get();
-+    default:
+}
-+      RTC_NOTREACHED();
+RTC_CHECK_NOTREACHED();
-+      return nullptr;
++    return nullptr;
-}
 }
 // Updates encoder with last observed parameters, such as callbacks, rates,
 // etc.
 void PrimeEncoder(VideoEncoder* encoder) const;
 // Settings used in the last InitEncode call and used if a dynamic fallback to
-@@ -334,16 +337,19 @@ int32_t VideoEncoderSoftwareFallbackWrap
+// software is required.
-case EncoderState::kUninitialized:
+@@ -338,16 +339,17 @@ int32_t VideoEncoderSoftwareFallbackWrap
-return WEBRTC_VIDEO_CODEC_ERROR;
 case EncoderState::kMainEncoderUsed: {
 return EncodeWithMainEncoder(frame, frame_types);
 }
 case EncoderState::kFallbackDueToFailure:
 case EncoderState::kForcedFallback:
 return fallback_encoder_->Encode(frame, frame_types);
-+    default:
-+      RTC_NOTREACHED();
-+      return WEBRTC_VIDEO_CODEC_ERROR;
 }
-}
+RTC_CHECK_NOTREACHED();
++  return WEBRTC_VIDEO_CODEC_ERROR;
+}
 int32_t VideoEncoderSoftwareFallbackWrapper::EncodeWithMainEncoder(
 const VideoFrame& frame,
 const std::vector<VideoFrameType>* frame_types) {
 int32_t ret = encoder_->Encode(frame, frame_types);
 // If requested, try a software fallback.
 bool fallback_requested = (ret == WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE);
 diff --git a/third_party/libwebrtc/call/adaptation/video_stream_adapter.cc b/third_party/libwebrtc/call/adaptation/video_stream_adapter.cc
 --- a/third_party/libwebrtc/call/adaptation/video_stream_adapter.cc
 +++ b/third_party/libwebrtc/call/adaptation/video_stream_adapter.cc
-@@ -156,16 +156,19 @@ const char* Adaptation::StatusToString(A
+@@ -163,16 +163,17 @@ const char* Adaptation::StatusToString(A
-case Adaptation::Status::kAwaitingPreviousAdaptation:
-return "kAwaitingPreviousAdaptation";
 case Status::kInsufficientInput:
 return "kInsufficientInput";
 case Status::kAdaptationDisabled:
 return "kAdaptationDisabled";
 case Status::kRejectedByConstraint:
 return "kRejectedByConstraint";
-+    default:
-+      RTC_NOTREACHED();
-+      return "";
 }
+RTC_CHECK_NOTREACHED();
++  return "";
 }
 Adaptation::Adaptation(int validation_id,
 VideoSourceRestrictions restrictions,
 VideoAdaptationCounters counters,
 VideoStreamInputState input_state)
 : validation_id_(validation_id),
-@@ -375,16 +378,19 @@ VideoStreamAdapter::RestrictionsOrState
+status_(Status::kValid),
-return IncreaseResolution(input_state, current_restrictions_);
+@@ -385,16 +386,17 @@ VideoStreamAdapter::RestrictionsOrState
-}
 case DegradationPreference::MAINTAIN_RESOLUTION: {
 // Scale up framerate.
 return IncreaseFramerate(input_state, current_restrictions_);
 }
 case DegradationPreference::DISABLED:
 return Adaptation::Status::kAdaptationDisabled;
-+    default:
-+      RTC_NOTREACHED();
-+      return Adaptation::Status::kAdaptationDisabled;
 }
+RTC_CHECK_NOTREACHED();
++  return Adaptation::Status::kAdaptationDisabled;
 }
 Adaptation VideoStreamAdapter::GetAdaptationDown() {
 RTC_DCHECK_RUN_ON(&sequence_checker_);
 VideoStreamInputState input_state = input_state_provider_->InputState();
 ++adaptation_validation_id_;
 RestrictionsOrState restrictions_or_state =
-@@ -454,16 +460,19 @@ VideoStreamAdapter::GetAdaptationDownSte
+GetAdaptationDownStep(input_state, current_restrictions_);
-case DegradationPreference::MAINTAIN_FRAMERATE: {
+@@ -467,16 +469,17 @@ VideoStreamAdapter::GetAdaptationDownSte
-return DecreaseResolution(input_state, current_restrictions);
 }
 case DegradationPreference::MAINTAIN_RESOLUTION: {
 return DecreaseFramerate(input_state, current_restrictions);
 }
 case DegradationPreference::DISABLED:
 return Adaptation::Status::kAdaptationDisabled;
-+    default:
-+      RTC_NOTREACHED();
-+      return Adaptation::Status::kAdaptationDisabled;
 }
+RTC_CHECK_NOTREACHED();
++  return Adaptation::Status::kAdaptationDisabled;
 }
 VideoStreamAdapter::RestrictionsOrState VideoStreamAdapter::DecreaseResolution(
 const VideoStreamInputState& input_state,
 const RestrictionsWithCounters& current_restrictions) {
 int target_pixels =
 GetLowerResolutionThan(input_state.frame_size_pixels().value());
-@@ -594,16 +603,18 @@ Adaptation VideoStreamAdapter::GetAdaptD
+// Use single active stream if set, this stream could be lower than the input.
+@@ -620,16 +623,18 @@ Adaptation VideoStreamAdapter::GetAdaptD
 case DegradationPreference::MAINTAIN_FRAMERATE:
 return GetAdaptationDown();
 case DegradationPreference::BALANCED: {
 return RestrictionsOrStateToAdaptation(
 GetAdaptDownResolutionStepForBalanced(input_state), input_state);
 }
-default:
-RTC_NOTREACHED();
-+      return RestrictionsOrStateToAdaptation(
-+          Adaptation::Status::kAdaptationDisabled, input_state);
 }
+RTC_CHECK_NOTREACHED();
++  return RestrictionsOrStateToAdaptation(
++         Adaptation::Status::kAdaptationDisabled, input_state);
 }
 VideoStreamAdapter::RestrictionsOrState
 VideoStreamAdapter::GetAdaptDownResolutionStepForBalanced(
 const VideoStreamInputState& input_state) const {
 // Adapt twice if the first adaptation did not decrease resolution.
 auto first_step = GetAdaptationDownStep(input_state, current_restrictions_);
+if (!absl::holds_alternative<RestrictionsWithCounters>(first_step)) {
 diff --git a/third_party/libwebrtc/call/simulated_network.cc b/third_party/libwebrtc/call/simulated_network.cc
 --- a/third_party/libwebrtc/call/simulated_network.cc
 +++ b/third_party/libwebrtc/call/simulated_network.cc
-@@ -72,16 +72,18 @@ bool CoDelSimulation::DropDequeuedPacket
+@@ -73,16 +73,17 @@ bool CoDelSimulation::DropDequeuedPacket
-if (queue_size - packet_size < kMaxPacketSize)
 state_ = kPending;
 last_drop_at_ = next_drop_at;
 ++drop_count_;
 return true;
 }
 return false;
 }
-+  RTC_NOTREACHED();
+RTC_CHECK_NOTREACHED();
 +  return false;
 }
 SimulatedNetwork::SimulatedNetwork(Config config, uint64_t random_seed)
 : random_(random_seed), bursting_(false) {
 SimulatedNetwork::~SimulatedNetwork() = default;
 diff --git a/third_party/libwebrtc/call/video_send_stream.cc b/third_party/libwebrtc/call/video_send_stream.cc
 --- a/third_party/libwebrtc/call/video_send_stream.cc
 +++ b/third_party/libwebrtc/call/video_send_stream.cc
-@@ -22,16 +22,19 @@ namespace {
+@@ -25,16 +25,17 @@ const char* StreamTypeToString(VideoSend
-const char* StreamTypeToString(VideoSendStream::StreamStats::StreamType type) {
-switch (type) {
 case VideoSendStream::StreamStats::StreamType::kMedia:
 return "media";
 case VideoSendStream::StreamStats::StreamType::kRtx:
 return "rtx";
 case VideoSendStream::StreamStats::StreamType::kFlexfec:
 return "flexfec";
-+    default:
-+        RTC_NOTREACHED();
-+        return "";
 }
+RTC_CHECK_NOTREACHED();
++  return "";
 }
 }  // namespace
 VideoSendStream::StreamStats::StreamStats() = default;
 VideoSendStream::StreamStats::~StreamStats() = default;
-diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc
+std::string VideoSendStream::StreamStats::ToString() const {
---- a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc
+diff --git a/third_party/libwebrtc/modules/audio_processing/agc/clipping_predictor.cc b/third_party/libwebrtc/modules/audio_processing/agc/clipping_predictor.cc
-+++ b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc
+--- a/third_party/libwebrtc/modules/audio_processing/agc/clipping_predictor.cc
-@@ -347,17 +347,17 @@ NetEq::Operation DecisionLogic::FuturePa
++++ b/third_party/libwebrtc/modules/audio_processing/agc/clipping_predictor.cc
-return NetEq::Operation::kNormal;
+@@ -373,11 +373,12 @@ std::unique_ptr<ClippingPredictor> Creat
+/*adaptive_step_estimation=*/true);
+case ClippingPredictorMode::kFixedStepClippingPeakPrediction:
+return std::make_unique<ClippingPeakPredictor>(
+num_channels, config.window_length, config.reference_window_length,
+config.reference_window_delay, config.clipping_threshold,
+/*adaptive_step_estimation=*/false);
 }
+RTC_DCHECK_NOTREACHED();
-// If previous was comfort noise, then no merge is needed.
++  return nullptr;
-if (prev_mode == NetEq::Mode::kRfc3389Cng ||
+}
-prev_mode == NetEq::Mode::kCodecInternalCng) {
-size_t cur_size_samples =
+}  // namespace webrtc
-estimate_dtx_delay_
+diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
--            ? cur_size_samples = span_samples_in_packet_buffer
+--- a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
-+            ? span_samples_in_packet_buffer
++++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
-: num_packets_in_packet_buffer * decoder_frame_length;
+@@ -54,16 +54,18 @@ std::vector<float> PreprocessWeights(rtc
-// Target level is in number of packets in Q8.
+rtc::FunctionView<float(float)> GetActivationFunction(
-const size_t target_level_samples =
+ActivationFunction activation_function) {
-(delay_manager_->TargetLevel() * packet_length_samples_) >> 8;
+switch (activation_function) {
-const bool generated_enough_noise =
+case ActivationFunction::kTansigApproximated:
-static_cast<uint32_t>(generated_noise_samples + target_timestamp) >=
+return ::rnnoise::TansigApproximated;
-available_timestamp;
+case ActivationFunction::kSigmoidApproximated:
+return ::rnnoise::SigmoidApproximated;
+}
++  // supposed to be never reached apparently therefore returning bogus
++  return ::rnnoise::TansigApproximated;
+}
+}  // namespace
+FullyConnectedLayer::FullyConnectedLayer(
+const int input_size,
+const int output_size,
+const rtc::ArrayView<const int8_t> bias,
 diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc
 --- a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc
 +++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc
-@@ -108,16 +108,19 @@ GainControl::Mode Agc1ConfigModeToInterf
+@@ -116,16 +116,17 @@ GainControl::Mode Agc1ConfigModeToInterf
-using Agc1Config = AudioProcessing::Config::GainController1;
-switch (mode) {
 case Agc1Config::kAdaptiveAnalog:
 return GainControl::kAdaptiveAnalog;
 case Agc1Config::kAdaptiveDigital:
 return GainControl::kAdaptiveDigital;
 case Agc1Config::kFixedDigital:
 return GainControl::kFixedDigital;
-+    default:
-+      RTC_NOTREACHED();
-+      return GainControl::kAdaptiveAnalog;
 }
+RTC_CHECK_NOTREACHED();
++  return GainControl::kAdaptiveAnalog;
+}
+bool MinimizeProcessingForUnusedOutput() {
+return !field_trial::IsEnabled("WebRTC-MutedStateKillSwitch");
 }
 // Maximum lengths that frame of samples being passed from the render side to
 // the capture side can have (does not apply to AEC3).
-static const size_t kMaxAllowedValuesOfSamplesPerBand = 160;
+@@ -1921,16 +1922,17 @@ void AudioProcessingImpl::InitializeNois
-static const size_t kMaxAllowedValuesOfSamplesPerFrame = 480;
-@@ -1847,16 +1850,17 @@ void AudioProcessingImpl::InitializeNois
 case NoiseSuppresionConfig::kModerate:
 return NsConfig::SuppressionLevel::k12dB;
 case NoiseSuppresionConfig::kHigh:
 return NsConfig::SuppressionLevel::k18dB;
 case NoiseSuppresionConfig::kVeryHigh:
 return NsConfig::SuppressionLevel::k21dB;
-default:
-RTC_NOTREACHED();
-+              return NsConfig::SuppressionLevel::k6dB;
 }
+RTC_CHECK_NOTREACHED();
++	  return NsConfig::SuppressionLevel::k6dB;
 };
 NsConfig cfg;
 cfg.target_level = map_level(config_.noise_suppression.level);
 submodules_.noise_suppressor = std::make_unique<NoiseSuppressor>(
 cfg, proc_sample_rate_hz(), num_proc_channels());
 }
+}
 diff --git a/third_party/libwebrtc/modules/audio_processing/include/audio_processing.cc b/third_party/libwebrtc/modules/audio_processing/include/audio_processing.cc
 --- a/third_party/libwebrtc/modules/audio_processing/include/audio_processing.cc
 +++ b/third_party/libwebrtc/modules/audio_processing/include/audio_processing.cc
-@@ -22,38 +22,47 @@ std::string NoiseSuppressionLevelToStrin
+@@ -27,28 +27,30 @@ std::string NoiseSuppressionLevelToStrin
-case AudioProcessing::Config::NoiseSuppression::Level::kLow:
-return "Low";
 case AudioProcessing::Config::NoiseSuppression::Level::kModerate:
 return "Moderate";
 case AudioProcessing::Config::NoiseSuppression::Level::kHigh:
 return "High";
 case AudioProcessing::Config::NoiseSuppression::Level::kVeryHigh:
 return "VeryHigh";
-+    default:
-+      RTC_NOTREACHED();
-+      return "";
 }
-}
+RTC_CHECK_NOTREACHED();
++  return "";
-std::string GainController1ModeToString(
+}
-const AudioProcessing::Config::GainController1::Mode& mode) {
+std::string GainController1ModeToString(const Agc1Config::Mode& mode) {
 switch (mode) {
-case AudioProcessing::Config::GainController1::Mode::kAdaptiveAnalog:
+case Agc1Config::Mode::kAdaptiveAnalog:
 return "AdaptiveAnalog";
-case AudioProcessing::Config::GainController1::Mode::kAdaptiveDigital:
+case Agc1Config::Mode::kAdaptiveDigital:
 return "AdaptiveDigital";
-case AudioProcessing::Config::GainController1::Mode::kFixedDigital:
+case Agc1Config::Mode::kFixedDigital:
 return "FixedDigital";
-+    default:
-+      RTC_NOTREACHED();
-+      return "";
 }
-}
+RTC_CHECK_NOTREACHED();
++  return "";
-std::string GainController2LevelEstimatorToString(
-const AudioProcessing::Config::GainController2::LevelEstimator& level) {
-switch (level) {
-case AudioProcessing::Config::GainController2::LevelEstimator::kRms:
-return "Rms";
-case AudioProcessing::Config::GainController2::LevelEstimator::kPeak:
-return "Peak";
-+    default:
-+      RTC_NOTREACHED();
-+      return "";
-}
-}
-int GetDefaultMaxInternalRate() {
-#ifdef WEBRTC_ARCH_ARM_FAMILY
-return 32000;
-#else
-return 48000;
-diff --git a/third_party/libwebrtc/modules/pacing/pacing_controller.cc b/third_party/libwebrtc/modules/pacing/pacing_controller.cc
---- a/third_party/libwebrtc/modules/pacing/pacing_controller.cc
-+++ b/third_party/libwebrtc/modules/pacing/pacing_controller.cc
-@@ -78,16 +78,19 @@ int GetPriorityForType(RtpPacketMediaTyp
-// Video has "normal" priority, in the old speak.
-// Send redundancy concurrently to video. If it is delayed it might have a
-// lower chance of being useful.
-return kFirstPriority + 3;
-case RtpPacketMediaType::kPadding:
-// Packets that are in themselves likely useless, only sent to keep the
-// BWE high.
-return kFirstPriority + 4;
-+    default:
-+      RTC_NOTREACHED();
-+      return -1;
-}
 }
 }  // namespace
-const TimeDelta PacingController::kMaxExpectedQueueLength =
+constexpr int AudioProcessing::kNativeSampleRatesHz[];
-TimeDelta::Millis(2000);
-const float PacingController::kDefaultPaceMultiplier = 2.5f;
+void CustomProcessing::SetRuntimeSetting(
+AudioProcessing::RuntimeSetting setting) {}
 diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/create_video_rtp_depacketizer.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/create_video_rtp_depacketizer.cc
 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/create_video_rtp_depacketizer.cc
 +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/create_video_rtp_depacketizer.cc
-@@ -31,12 +31,15 @@ std::unique_ptr<VideoRtpDepacketizer> Cr
+@@ -33,11 +33,12 @@ std::unique_ptr<VideoRtpDepacketizer> Cr
-return std::make_unique<VideoRtpDepacketizerVp8>();
-case kVideoCodecVP9:
 return std::make_unique<VideoRtpDepacketizerVp9>();
 case kVideoCodecAV1:
 return std::make_unique<VideoRtpDepacketizerAv1>();
 case kVideoCodecGeneric:
 case kVideoCodecMultiplex:
 return std::make_unique<VideoRtpDepacketizerGeneric>();
-+    default:
-+      RTC_NOTREACHED();
-+      return nullptr;
 }
+RTC_CHECK_NOTREACHED();
++  return nullptr;
 }
 }  // namespace webrtc
 diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.cc
 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.cc
 +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.cc
-@@ -125,16 +125,19 @@ bool IsNonVolatile(RTPExtensionType type
+@@ -132,16 +132,17 @@ bool IsNonVolatile(RTPExtensionType type
-case kRtpExtensionNone:
+#if defined(WEBRTC_MOZILLA_BUILD)
-case kRtpExtensionNumberOfExtensions:
-RTC_NOTREACHED();
-return false;
 case kRtpExtensionCsrcAudioLevel:
 // TODO: Mozilla implement for CsrcAudioLevel
 RTC_CHECK(false);
 return false;
-+    default:
+#endif
-+      RTC_NOTREACHED();
-+      return false;
 }
+RTC_CHECK_NOTREACHED();
++  return false;
 }
 bool HasBweExtension(const RtpHeaderExtensionMap& extensions_map) {
 return extensions_map.IsRegistered(kRtpExtensionTransportSequenceNumber) ||
 extensions_map.IsRegistered(kRtpExtensionTransportSequenceNumber02) ||
 extensions_map.IsRegistered(kRtpExtensionAbsoluteSendTime) ||
 extensions_map.IsRegistered(kRtpExtensionTransmissionTimeOffset);
+}
 diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
 --- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
 +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
-@@ -40,16 +40,19 @@ namespace {
+@@ -42,16 +42,17 @@ const char* FrameTypeToString(AudioFrame
-const char* FrameTypeToString(AudioFrameType frame_type) {
-switch (frame_type) {
 case AudioFrameType::kEmptyFrame:
 return "empty";
 case AudioFrameType::kAudioFrameSpeech:
 return "audio_speech";
 case AudioFrameType::kAudioFrameCN:
 return "audio_cn";
-+    default:
-+      RTC_NOTREACHED();
-+      return "";
 }
+RTC_CHECK_NOTREACHED();
++  return "";
 }
 #endif
 constexpr char kIncludeCaptureClockOffset[] =
 "WebRTC-IncludeCaptureClockOffset";
 }  // namespace
 diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/temporal_layers_checker.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp8/temporal_layers_checker.cc
 --- a/third_party/libwebrtc/modules/video_coding/codecs/vp8/temporal_layers_checker.cc
 +++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/temporal_layers_checker.cc
-@@ -23,16 +23,19 @@ TemporalLayersChecker::CreateTemporalLay
+@@ -25,16 +25,17 @@ TemporalLayersChecker::CreateTemporalLay
-int num_temporal_layers) {
-switch (type) {
 case Vp8TemporalLayersType::kFixedPattern:
 return std::make_unique<DefaultTemporalLayersChecker>(
 num_temporal_layers);
 case Vp8TemporalLayersType::kBitrateDynamic:
 // Conference mode temporal layering for screen content in base stream.
 return std::make_unique<TemporalLayersChecker>(num_temporal_layers);
-+    default:
-+      RTC_NOTREACHED();
-+      return nullptr;
 }
+RTC_CHECK_NOTREACHED();
++  return nullptr;
 }
 TemporalLayersChecker::TemporalLayersChecker(int num_temporal_layers)
 : num_temporal_layers_(num_temporal_layers),
 sequence_number_(0),
 last_sync_sequence_number_(0),
 last_tl0_sequence_number_(0) {}
 diff --git a/third_party/libwebrtc/video/adaptation/video_stream_encoder_resource_manager.cc b/third_party/libwebrtc/video/adaptation/video_stream_encoder_resource_manager.cc
 --- a/third_party/libwebrtc/video/adaptation/video_stream_encoder_resource_manager.cc
 +++ b/third_party/libwebrtc/video/adaptation/video_stream_encoder_resource_manager.cc
-@@ -49,16 +49,19 @@ bool IsFramerateScalingEnabled(Degradati
+@@ -58,16 +58,17 @@ bool IsFramerateScalingEnabled(Degradati
-}
 std::string ToString(VideoAdaptationReason reason) {
 switch (reason) {
 case VideoAdaptationReason::kQuality:
 return "quality";
 case VideoAdaptationReason::kCpu:
 return "cpu";
-+    default:
-+      RTC_NOTREACHED();
-+      return "";
 }
-}
+RTC_CHECK_NOTREACHED();
++  return "";
-}  // namespace
+}
-class VideoStreamEncoderResourceManager::InitialFrameDropper {
+std::vector<bool> GetActiveLayersFlags(const VideoCodec& codec) {
-public:
+std::vector<bool> flags;
-explicit InitialFrameDropper(
+if (codec.codecType == VideoCodecType::kVideoCodecVP9) {
+flags.resize(codec.VP9().numberOfSpatialLayers);
+for (size_t i = 0; i < flags.size(); ++i) {
+flags[i] = codec.spatialLayers[i].active;
 diff --git a/third_party/rust/glslopt/glsl-optimizer/src/compiler/glsl/ast_to_hir.cpp b/third_party/rust/glslopt/glsl-optimizer/src/compiler/glsl/ast_to_hir.cpp
 --- a/third_party/rust/glslopt/glsl-optimizer/src/compiler/glsl/ast_to_hir.cpp
 +++ b/third_party/rust/glslopt/glsl-optimizer/src/compiler/glsl/ast_to_hir.cpp
 @@ -2646,16 +2646,17 @@ get_type_name_for_precision_qualifier(co
 unreachable("Unsupported sampler/image type");

branch	firefox106
changeset 1180	d76083122710
parent 1179	c19c9e7820ef
child 1181	ba646dddffef