--- a/mozilla-silence-no-return-type.patch Mon Oct 17 21:08:02 2022 +0200
+++ b/mozilla-silence-no-return-type.patch Tue Nov 15 15:11:07 2022 +0100
@@ -1,10 +1,10 @@
# 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",
@@ -26,7 +26,7 @@
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" }
@@ -38,11 +38,11 @@
+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" }
- sql-support = { git = "https://github.com/mozilla/application-services", rev = "2689788cecf24c385e6b7440e3aa1a89c511f14a" }
- sync15-traits = { git = "https://github.com/mozilla/application-services", rev = "2689788cecf24c385e6b7440e3aa1a89c511f14a" }
- viaduct = { git = "https://github.com/mozilla/application-services", rev = "2689788cecf24c385e6b7440e3aa1a89c511f14a" }
- webext-storage = { 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 = "fb1c78b13c27b5db1fd5458b8c2d8f433855dd61" }
+ sync15-traits = { git = "https://github.com/mozilla/application-services", rev = "fb1c78b13c27b5db1fd5458b8c2d8f433855dd61" }
+ viaduct = { git = "https://github.com/mozilla/application-services", rev = "fb1c78b13c27b5db1fd5458b8c2d8f433855dd61" }
+ 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
@@ -1943,29 +1943,16 @@
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 @@
- * 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 {
-
+@@ -17,16 +17,17 @@ 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() {}
@@ -1973,22 +1960,20 @@
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
- case DegradationPreference::DISABLED:
- return "disabled";
+@@ -27,16 +27,17 @@ const char* DegradationPreferenceToStrin
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;
@@ -1996,22 +1981,42 @@
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
- case kVideoCodecAV1:
- return kPayloadNameAv1;
+@@ -117,16 +117,17 @@ const char* CodecTypeToPayloadString(Vid
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) {
@@ -2019,22 +2024,20 @@
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
- << "Trying to access encoder in uninitialized fallback wrapper.";
- // Return main encoder to preserve previous behavior.
- ABSL_FALLTHROUGH_INTENDED;
+@@ -158,16 +158,17 @@ class VideoEncoderSoftwareFallbackWrappe
+ [[fallthrough]];
case EncoderState::kMainEncoderUsed:
return encoder_.get();
case EncoderState::kFallbackDueToFailure:
case EncoderState::kForcedFallback:
return fallback_encoder_.get();
-+ default:
-+ RTC_NOTREACHED();
-+ return nullptr;
}
+ RTC_CHECK_NOTREACHED();
++ return nullptr;
}
// Updates encoder with last observed parameters, such as callbacks, rates,
@@ -2042,20 +2045,19 @@
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
- case EncoderState::kUninitialized:
- return WEBRTC_VIDEO_CODEC_ERROR;
+ // software is required.
+@@ -338,16 +339,17 @@ int32_t VideoEncoderSoftwareFallbackWrap
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) {
@@ -2065,19 +2067,16 @@
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
- case Adaptation::Status::kAwaitingPreviousAdaptation:
- return "kAwaitingPreviousAdaptation";
+@@ -163,16 +163,17 @@ const char* Adaptation::StatusToString(A
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,
@@ -2085,19 +2084,17 @@
VideoAdaptationCounters counters,
VideoStreamInputState input_state)
: validation_id_(validation_id),
-@@ -375,16 +378,19 @@ VideoStreamAdapter::RestrictionsOrState
- return IncreaseResolution(input_state, current_restrictions_);
- }
+ status_(Status::kValid),
+@@ -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() {
@@ -2105,19 +2102,17 @@
VideoStreamInputState input_state = input_state_provider_->InputState();
++adaptation_validation_id_;
RestrictionsOrState restrictions_or_state =
-@@ -454,16 +460,19 @@ VideoStreamAdapter::GetAdaptationDownSte
- case DegradationPreference::MAINTAIN_FRAMERATE: {
- return DecreaseResolution(input_state, current_restrictions);
+ GetAdaptationDownStep(input_state, current_restrictions_);
+@@ -467,16 +469,17 @@ VideoStreamAdapter::GetAdaptationDownSte
}
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(
@@ -2125,18 +2120,18 @@
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
@@ -2144,11 +2139,11 @@
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
- if (queue_size - packet_size < kMaxPacketSize)
+@@ -73,16 +73,17 @@ bool CoDelSimulation::DropDequeuedPacket
state_ = kPending;
last_drop_at_ = next_drop_at;
++drop_count_;
@@ -2156,7 +2151,7 @@
}
return false;
}
-+ RTC_NOTREACHED();
+ RTC_CHECK_NOTREACHED();
+ return false;
}
@@ -2169,19 +2164,16 @@
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 {
- const char* StreamTypeToString(VideoSendStream::StreamStats::StreamType type) {
- switch (type) {
+@@ -25,16 +25,17 @@ const char* StreamTypeToString(VideoSend
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
@@ -2189,62 +2181,76 @@
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
---- a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc
-+++ b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc
-@@ -347,17 +347,17 @@ NetEq::Operation DecisionLogic::FuturePa
- return NetEq::Operation::kNormal;
+ std::string VideoSendStream::StreamStats::ToString() const {
+diff --git a/third_party/libwebrtc/modules/audio_processing/agc/clipping_predictor.cc b/third_party/libwebrtc/modules/audio_processing/agc/clipping_predictor.cc
+--- a/third_party/libwebrtc/modules/audio_processing/agc/clipping_predictor.cc
++++ b/third_party/libwebrtc/modules/audio_processing/agc/clipping_predictor.cc
+@@ -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();
++ return nullptr;
+ }
- // If previous was comfort noise, then no merge is needed.
- if (prev_mode == NetEq::Mode::kRfc3389Cng ||
- prev_mode == NetEq::Mode::kCodecInternalCng) {
- size_t cur_size_samples =
- estimate_dtx_delay_
-- ? cur_size_samples = span_samples_in_packet_buffer
-+ ? span_samples_in_packet_buffer
- : num_packets_in_packet_buffer * decoder_frame_length;
- // Target level is in number of packets in Q8.
- const size_t target_level_samples =
- (delay_manager_->TargetLevel() * packet_length_samples_) >> 8;
- const bool generated_enough_noise =
- static_cast<uint32_t>(generated_noise_samples + target_timestamp) >=
- available_timestamp;
+ } // namespace webrtc
+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
+--- 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
+@@ -54,16 +54,18 @@ std::vector<float> PreprocessWeights(rtc
+ rtc::FunctionView<float(float)> GetActivationFunction(
+ ActivationFunction activation_function) {
+ switch (activation_function) {
+ case ActivationFunction::kTansigApproximated:
+ return ::rnnoise::TansigApproximated;
+ 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
- using Agc1Config = AudioProcessing::Config::GainController1;
- switch (mode) {
+@@ -116,16 +116,17 @@ GainControl::Mode Agc1ConfigModeToInterf
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;
- static const size_t kMaxAllowedValuesOfSamplesPerFrame = 480;
-
-@@ -1847,16 +1850,17 @@ void AudioProcessingImpl::InitializeNois
+@@ -1921,16 +1922,17 @@ 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;
@@ -2252,115 +2258,70 @@
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
- case AudioProcessing::Config::NoiseSuppression::Level::kLow:
- return "Low";
+@@ -27,28 +27,30 @@ std::string NoiseSuppressionLevelToStrin
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 "";
}
- }
-
- std::string GainController1ModeToString(
- const AudioProcessing::Config::GainController1::Mode& mode) {
- switch (mode) {
- case AudioProcessing::Config::GainController1::Mode::kAdaptiveAnalog:
- return "AdaptiveAnalog";
- case AudioProcessing::Config::GainController1::Mode::kAdaptiveDigital:
- return "AdaptiveDigital";
- case AudioProcessing::Config::GainController1::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 "";
+ std::string GainController1ModeToString(const Agc1Config::Mode& mode) {
+ switch (mode) {
+ case Agc1Config::Mode::kAdaptiveAnalog:
+ return "AdaptiveAnalog";
+ case Agc1Config::Mode::kAdaptiveDigital:
+ return "AdaptiveDigital";
+ case Agc1Config::Mode::kFixedDigital:
+ return "FixedDigital";
}
- }
-
- 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;
- }
+ RTC_CHECK_NOTREACHED();
++ return "";
}
} // namespace
- const TimeDelta PacingController::kMaxExpectedQueueLength =
- TimeDelta::Millis(2000);
- const float PacingController::kDefaultPaceMultiplier = 2.5f;
+ constexpr int AudioProcessing::kNativeSampleRatesHz[];
+
+ 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
- return std::make_unique<VideoRtpDepacketizerVp8>();
- case kVideoCodecVP9:
+@@ -33,11 +33,12 @@ std::unique_ptr<VideoRtpDepacketizer> Cr
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
- case kRtpExtensionNone:
- case kRtpExtensionNumberOfExtensions:
- RTC_NOTREACHED();
- return false;
+@@ -132,16 +132,17 @@ bool IsNonVolatile(RTPExtensionType type
+ #if defined(WEBRTC_MOZILLA_BUILD)
case kRtpExtensionCsrcAudioLevel:
// TODO: Mozilla implement for CsrcAudioLevel
RTC_CHECK(false);
return false;
-+ default:
-+ RTC_NOTREACHED();
-+ return false;
+ #endif
}
+ RTC_CHECK_NOTREACHED();
++ return false;
}
bool HasBweExtension(const RtpHeaderExtensionMap& extensions_map) {
@@ -2368,22 +2329,20 @@
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 {
- const char* FrameTypeToString(AudioFrameType frame_type) {
- switch (frame_type) {
+@@ -42,16 +42,17 @@ const char* FrameTypeToString(AudioFrame
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
@@ -2391,22 +2350,20 @@
"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
- int num_temporal_layers) {
- switch (type) {
+@@ -25,16 +25,17 @@ TemporalLayersChecker::CreateTemporalLay
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)
@@ -2414,29 +2371,28 @@
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 {
- public:
- explicit InitialFrameDropper(
+ std::vector<bool> GetActiveLayersFlags(const VideoCodec& codec) {
+ std::vector<bool> flags;
+ 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