termux-packages/x11-packages/electron-host-tools-for-code-oss/jumbo-patches/0115-webrtc.patch

From a70d36bc8930357ae55ca5e0932888833d0db9f4 Mon Sep 17 00:00:00 2001
From: Chongyun Lee <licy183@termux.dev>
Date: Sat, 23 Aug 2025 23:52:42 +0800
Subject: [PATCH] reland jumbo 15

Enable jumbo build for the following template(s):

- //third_party/webrtc/webrtc.gni -> template("rtc_source_set")
- //third_party/webrtc/webrtc.gni -> template("rtc_static_library")

---
 .../audio_encoder_multi_channel_opus_config.cc |  4 ++++
 .../opus/audio_encoder_opus_config.cc          |  4 ++++
 third_party/webrtc/api/video/i010_buffer.cc    |  6 ++++++
 third_party/webrtc/api/video/i210_buffer.cc    |  6 ++++++
 third_party/webrtc/api/video/i410_buffer.cc    |  6 ++++++
 third_party/webrtc/api/video/i420_buffer.cc    |  4 ++++
 third_party/webrtc/api/video/i422_buffer.cc    |  4 ++++
 .../api/video_codecs/h264_profile_level_id.cc  |  6 ++++++
 .../video_codecs/h265_profile_tier_level.cc    |  6 ++++++
 third_party/webrtc/common_audio/BUILD.gn       |  4 ++++
 .../common_video/h265/h265_bitstream_parser.cc |  6 ++++++
 .../common_video/h265/h265_pps_parser.cc       |  6 ++++++
 .../webrtc/media/engine/webrtc_video_engine.cc |  4 ++++
 .../webrtc/media/engine/webrtc_voice_engine.cc |  4 ++++
 .../audio_coding/neteq/reorder_optimizer.cc    |  6 ++++++
 .../audio_coding/neteq/underrun_optimizer.cc   |  6 ++++++
 .../remote_bitrate_estimator_abs_send_time.cc  |  6 ++++++
 .../remote_bitrate_estimator_single_stream.cc  |  6 ++++++
 ...flexfec_03_header_reader_writer_unittest.cc | 18 ++++++++++++++++++
 .../source/flexfec_header_reader_writer.cc     | 18 ++++++++++++++++++
 .../rtp_rtcp/source/flexfec_receiver.cc        |  4 ++++
 .../modules/rtp_rtcp/source/flexfec_sender.cc  |  4 ++++
 .../rtp_rtcp/source/packet_sequencer.cc        |  6 ++++++
 .../modules/rtp_rtcp/source/rtcp_receiver.cc   |  6 ++++++
 .../modules/rtp_rtcp/source/rtcp_sender.cc     |  6 ++++++
 .../modules/rtp_rtcp/source/rtp_packet.cc      |  4 ++++
 .../rtp_rtcp/source/rtp_sender_egress.cc       |  4 ++++
 .../rtp_rtcp/source/rtp_sender_video.cc        |  4 ++++
 .../webrtc/modules/rtp_rtcp/source/rtp_util.cc |  4 ++++
 .../codecs/vp8/screenshare_layers.cc           | 18 ++++++++++++++----
 .../scalability_structure_l2t2_key_shift.cc    |  4 ++++
 .../svc/scalability_structure_simulcast.cc     |  4 ++++
 .../webrtc/pc/jsep_session_description.cc      |  6 ++++++
 third_party/webrtc/pc/webrtc_sdp.cc            |  6 ++++++
 third_party/webrtc/webrtc.gni                  |  9 +++++----
 35 files changed, 211 insertions(+), 8 deletions(-)

diff --git a/third_party/webrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc b/third_party/webrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc
index a8123cb17a..662c99eae9 100644
--- a/third_party/webrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc
+++ b/third_party/webrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc
@@ -15,6 +15,8 @@
 
 #include "api/audio_codecs/audio_encoder.h"
 
+#define kDefaultComplexity kDefaultComplexity_AudioEncoderMultiChannelOpusConfig
+
 namespace webrtc {
 
 namespace {
@@ -110,3 +112,5 @@ bool AudioEncoderMultiChannelOpusConfig::IsOk() const {
 }
 
 }  // namespace webrtc
+
+#undef kDefaultComplexity
diff --git a/third_party/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc b/third_party/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc
index 92ee7bd66c..18f5529732 100644
--- a/third_party/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc
+++ b/third_party/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc
@@ -12,6 +12,8 @@
 
 #include "api/audio_codecs/audio_encoder.h"
 
+#define kDefaultComplexity kDefaultComplexity_AudioEncoderOpusConfig
+
 namespace webrtc {
 
 namespace {
@@ -75,3 +77,5 @@ bool AudioEncoderOpusConfig::IsOk() const {
   return true;
 }
 }  // namespace webrtc
+
+#undef kDefaultComplexity
diff --git a/third_party/webrtc/api/video/i010_buffer.cc b/third_party/webrtc/api/video/i010_buffer.cc
index 59da7fb114..b1266e0453 100644
--- a/third_party/webrtc/api/video/i010_buffer.cc
+++ b/third_party/webrtc/api/video/i010_buffer.cc
@@ -25,6 +25,9 @@
 #include "third_party/libyuv/include/libyuv/rotate.h"
 #include "third_party/libyuv/include/libyuv/scale.h"
 
+#define kBufferAlignment kBufferAlignment_I010Buffer
+#define kBytesPerPixel   kBytesPerPixel_I010Buffer
+
 // Aligning pointer to 64 bytes for improved performance, e.g. use SIMD.
 static const int kBufferAlignment = 64;
 static const int kBytesPerPixel = 2;
@@ -215,3 +218,6 @@ void I010Buffer::ScaleFrom(const I010BufferInterface& src) {
 }
 
 }  // namespace webrtc
+
+#undef kBufferAlignment
+#undef kBytesPerPixel
diff --git a/third_party/webrtc/api/video/i210_buffer.cc b/third_party/webrtc/api/video/i210_buffer.cc
index 1f7919e3b4..41f087d7a9 100644
--- a/third_party/webrtc/api/video/i210_buffer.cc
+++ b/third_party/webrtc/api/video/i210_buffer.cc
@@ -26,6 +26,9 @@
 #include "third_party/libyuv/include/libyuv/rotate.h"
 #include "third_party/libyuv/include/libyuv/scale.h"
 
+#define kBufferAlignment kBufferAlignment_I210Buffer
+#define kBytesPerPixel   kBytesPerPixel_I210Buffer
+
 // Aligning pointer to 64 bytes for improved performance, e.g. use SIMD.
 static const int kBufferAlignment = 64;
 static const int kBytesPerPixel = 2;
@@ -214,3 +217,6 @@ void I210Buffer::ScaleFrom(const I210BufferInterface& src) {
 }
 
 }  // namespace webrtc
+
+#undef kBufferAlignment
+#undef kBytesPerPixel
diff --git a/third_party/webrtc/api/video/i410_buffer.cc b/third_party/webrtc/api/video/i410_buffer.cc
index 37564b9f3a..d500f56dfa 100644
--- a/third_party/webrtc/api/video/i410_buffer.cc
+++ b/third_party/webrtc/api/video/i410_buffer.cc
@@ -28,6 +28,9 @@
 #include "third_party/libyuv/include/libyuv/rotate.h"
 #include "third_party/libyuv/include/libyuv/scale.h"
 
+#define kBufferAlignment kBufferAlignment_I410Buffer
+#define kBytesPerPixel   kBytesPerPixel_I410Buffer
+
 // Aligning pointer to 64 bytes for improved performance, e.g. use SIMD.
 static const int kBufferAlignment = 64;
 static const int kBytesPerPixel = 2;
@@ -224,3 +227,6 @@ void I410Buffer::ScaleFrom(const I410BufferInterface& src) {
 }
 
 }  // namespace webrtc
+
+#undef kBufferAlignment
+#undef kBytesPerPixel
diff --git a/third_party/webrtc/api/video/i420_buffer.cc b/third_party/webrtc/api/video/i420_buffer.cc
index 33b2ea666e..28a1fcad3e 100644
--- a/third_party/webrtc/api/video/i420_buffer.cc
+++ b/third_party/webrtc/api/video/i420_buffer.cc
@@ -27,6 +27,8 @@
 #include "third_party/libyuv/include/libyuv/rotate.h"
 #include "third_party/libyuv/include/libyuv/scale.h"
 
+#define kBufferAlignment kBufferAlignment_i420
+
 // Aligning pointer to 64 bytes for improved performance, e.g. use SIMD.
 static const int kBufferAlignment = 64;
 
@@ -239,3 +241,5 @@ void I420Buffer::ScaleFrom(const I420BufferInterface& src) {
 }
 
 }  // namespace webrtc
+
+#undef kBufferAlignment
diff --git a/third_party/webrtc/api/video/i422_buffer.cc b/third_party/webrtc/api/video/i422_buffer.cc
index 25814a88d9..b6016311c2 100644
--- a/third_party/webrtc/api/video/i422_buffer.cc
+++ b/third_party/webrtc/api/video/i422_buffer.cc
@@ -29,6 +29,8 @@
 #include "third_party/libyuv/include/libyuv/rotate.h"
 #include "third_party/libyuv/include/libyuv/scale.h"
 
+#define kBufferAlignment kBufferAlignment_i422
+
 // Aligning pointer to 64 bytes for improved performance, e.g. use SIMD.
 static const int kBufferAlignment = 64;
 
@@ -241,3 +243,5 @@ void I422Buffer::CropAndScaleFrom(const I422BufferInterface& src,
 }
 
 }  // namespace webrtc
+
+#undef kBufferAlignment
diff --git a/third_party/webrtc/api/video_codecs/h264_profile_level_id.cc b/third_party/webrtc/api/video_codecs/h264_profile_level_id.cc
index e410fc22f2..433fcc5800 100644
--- a/third_party/webrtc/api/video_codecs/h264_profile_level_id.cc
+++ b/third_party/webrtc/api/video_codecs/h264_profile_level_id.cc
@@ -19,6 +19,9 @@
 
 #include "api/rtp_parameters.h"
 
+#define LevelConstraint   LevelConstraint_H264ProfileLevelId
+#define kLevelConstraints kLevelConstraints_H264ProfileLevelId
+
 namespace webrtc {
 
 namespace {
@@ -269,3 +272,6 @@ bool H264IsSameProfileAndLevel(const CodecParameterMap& params1,
 }
 
 }  // namespace webrtc
+
+#undef LevelConstraint
+#undef kLevelConstraints
diff --git a/third_party/webrtc/api/video_codecs/h265_profile_tier_level.cc b/third_party/webrtc/api/video_codecs/h265_profile_tier_level.cc
index 4ec1c8070f..427f404c06 100644
--- a/third_party/webrtc/api/video_codecs/h265_profile_tier_level.cc
+++ b/third_party/webrtc/api/video_codecs/h265_profile_tier_level.cc
@@ -17,6 +17,9 @@
 #include "api/video/resolution.h"
 #include "rtc_base/string_to_number.h"
 
+#define LevelConstraint   LevelConstraint_H265ProfileTierLevel
+#define kLevelConstraints kLevelConstraints_H265ProfileTierLevel
+
 namespace webrtc {
 
 namespace {
@@ -324,3 +327,6 @@ std::optional<H265Level> GetSupportedH265Level(const Resolution& resolution,
 }
 
 }  // namespace webrtc
+
+#undef LevelConstraint
+#undef kLevelConstraints
diff --git a/third_party/webrtc/common_audio/BUILD.gn b/third_party/webrtc/common_audio/BUILD.gn
index c5459b21bb..b39d9d2761 100644
--- a/third_party/webrtc/common_audio/BUILD.gn
+++ b/third_party/webrtc/common_audio/BUILD.gn
@@ -41,6 +41,10 @@ rtc_library("common_audio") {
     "window_generator.cc",
     "window_generator.h",
   ]
+  jumbo_excluded_sources = [
+    # Use `#pragma pack`
+    "wav_header.cc",
+  ]
 
   deps = [
     ":common_audio_c",
diff --git a/third_party/webrtc/common_video/h265/h265_bitstream_parser.cc b/third_party/webrtc/common_video/h265/h265_bitstream_parser.cc
index b3f9793a2a..30d27cd3de 100644
--- a/third_party/webrtc/common_video/h265/h265_bitstream_parser.cc
+++ b/third_party/webrtc/common_video/h265/h265_bitstream_parser.cc
@@ -25,6 +25,8 @@
 #include "rtc_base/bitstream_reader.h"
 #include "rtc_base/logging.h"
 
+#define kMaxRefIdxActive kMaxRefIdxActive_H265BitstreamParser
+
 #define IN_RANGE_OR_RETURN(val, min, max)                                     \
   do {                                                                        \
     if (!slice_reader.Ok() || (val) < (min) || (val) > (max)) {               \
@@ -668,3 +670,7 @@ std::optional<uint32_t> H265BitstreamParser::GetLastSlicePpsId() const {
 }
 
 }  // namespace webrtc
+
+#undef kMaxRefIdxActive
+#undef IN_RANGE_OR_RETURN_NULL
+#undef TRUE_OR_RETURN
diff --git a/third_party/webrtc/common_video/h265/h265_pps_parser.cc b/third_party/webrtc/common_video/h265/h265_pps_parser.cc
index c94e362ec9..ee22d80b1c 100644
--- a/third_party/webrtc/common_video/h265/h265_pps_parser.cc
+++ b/third_party/webrtc/common_video/h265/h265_pps_parser.cc
@@ -21,6 +21,8 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 
+#define kMaxRefIdxActive kMaxRefIdxActive_H265PpsParser
+
 #define IN_RANGE_OR_RETURN_NULL(val, min, max)                                \
   do {                                                                        \
     if (!reader.Ok() || (val) < (min) || (val) > (max)) {                     \
@@ -249,3 +251,7 @@ bool H265PpsParser::ParsePpsIdsInternal(BitstreamReader& reader,
 }
 
 }  // namespace webrtc
+
+#undef kMaxRefIdxActive
+#undef IN_RANGE_OR_RETURN_NULL
+#undef TRUE_OR_RETURN
diff --git a/third_party/webrtc/media/engine/webrtc_video_engine.cc b/third_party/webrtc/media/engine/webrtc_video_engine.cc
index 9b71d020df..c9eb8bd027 100644
--- a/third_party/webrtc/media/engine/webrtc_video_engine.cc
+++ b/third_party/webrtc/media/engine/webrtc_video_engine.cc
@@ -101,6 +101,8 @@
 #include "rtc_base/trace_event.h"
 #include "video/config/video_encoder_config.h"
 
+#define ValidateStreamParams ValidateStreamParams_WebRTCVideoEngine
+
 namespace webrtc {
 
 namespace {
@@ -4095,3 +4097,5 @@ bool VideoCodecSettings::operator!=(const VideoCodecSettings& other) const {
 }
 
 }  // namespace webrtc
+
+#undef ValidateStreamParams
diff --git a/third_party/webrtc/media/engine/webrtc_voice_engine.cc b/third_party/webrtc/media/engine/webrtc_voice_engine.cc
index 3690ffb255..d3364705fd 100644
--- a/third_party/webrtc/media/engine/webrtc_voice_engine.cc
+++ b/third_party/webrtc/media/engine/webrtc_voice_engine.cc
@@ -116,6 +116,8 @@
 #include "api/audio/create_audio_device_module.h"
 #endif
 
+#define ValidateStreamParams ValidateStreamParams_WebRTCVoiceEngine
+
 namespace webrtc {
 namespace {
 
@@ -2808,3 +2810,5 @@ bool WebRtcVoiceReceiveChannel::MaybeDeregisterUnsignaledRecvStream(
   return false;
 }
 }  // namespace webrtc
+
+#undef ValidateStreamParams
diff --git a/third_party/webrtc/modules/audio_coding/neteq/reorder_optimizer.cc b/third_party/webrtc/modules/audio_coding/neteq/reorder_optimizer.cc
index 98dbedfaa4..466ac1c308 100644
--- a/third_party/webrtc/modules/audio_coding/neteq/reorder_optimizer.cc
+++ b/third_party/webrtc/modules/audio_coding/neteq/reorder_optimizer.cc
@@ -17,6 +17,9 @@
 #include <optional>
 #include <vector>
 
+#define kDelayBuckets kDelayBuckets_ReorderOptimizer
+#define kBucketSizeMs kBucketSizeMs_ReorderOptimizer
+
 namespace webrtc {
 
 namespace {
@@ -76,3 +79,6 @@ int ReorderOptimizer::MinimizeCostFunction(int base_delay_ms) const {
 }
 
 }  // namespace webrtc
+
+#undef kDelayBuckets
+#undef kBucketSizeMs
diff --git a/third_party/webrtc/modules/audio_coding/neteq/underrun_optimizer.cc b/third_party/webrtc/modules/audio_coding/neteq/underrun_optimizer.cc
index dcee8e19fe..045d5137e4 100644
--- a/third_party/webrtc/modules/audio_coding/neteq/underrun_optimizer.cc
+++ b/third_party/webrtc/modules/audio_coding/neteq/underrun_optimizer.cc
@@ -15,6 +15,9 @@
 
 #include "api/neteq/tick_timer.h"
 
+#define kDelayBuckets kDelayBuckets_UnderrunOptimizer
+#define kBucketSizeMs kBucketSizeMs_UnderrunOptimizer
+
 namespace webrtc {
 
 namespace {
@@ -72,3 +75,6 @@ void UnderrunOptimizer::Reset() {
 }
 
 }  // namespace webrtc
+
+#undef kDelayBuckets
+#undef kBucketSizeMs
diff --git a/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc b/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
index 5847b088b9..0c0209394c 100644
--- a/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
+++ b/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
@@ -38,6 +38,9 @@
 #include "rtc_base/logging.h"
 #include "system_wrappers/include/metrics.h"
 
+#define kTimestampGroupLengthMs kTimestampGroupLengthMs_RemoteBitrateEstimatorAbsSendTime
+#define kTimestampToMs          kTimestampToMs_RemoteBitrateEstimatorAbsSendTime
+
 namespace webrtc {
 namespace {
 
@@ -376,3 +379,6 @@ DataRate RemoteBitrateEstimatorAbsSendTime::LatestEstimate() const {
 }
 
 }  // namespace webrtc
+
+#undef kTimestampGroupLengthMs
+#undef kTimestampToMs
diff --git a/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc b/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
index b05610d4e9..7ce25c3a2a 100644
--- a/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
+++ b/third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
@@ -34,6 +34,9 @@
 #include "system_wrappers/include/clock.h"
 #include "system_wrappers/include/metrics.h"
 
+#define kTimestampGroupLengthMs kTimestampGroupLengthMs_RemoteBitrateEstimatorSingleStream
+#define kTimestampToMs          kTimestampToMs_RemoteBitrateEstimatorSingleStream
+
 namespace webrtc {
 namespace {
 
@@ -195,3 +198,6 @@ std::vector<uint32_t> RemoteBitrateEstimatorSingleStream::GetSsrcs() const {
 }
 
 }  // namespace webrtc
+
+#undef kTimestampGroupLengthMs
+#undef kTimestampToMs
diff --git a/third_party/webrtc/modules/rtp_rtcp/source/flexfec_03_header_reader_writer_unittest.cc b/third_party/webrtc/modules/rtp_rtcp/source/flexfec_03_header_reader_writer_unittest.cc
index 80b4ba3373..a0e9b2e39a 100644
--- a/third_party/webrtc/modules/rtp_rtcp/source/flexfec_03_header_reader_writer_unittest.cc
+++ b/third_party/webrtc/modules/rtp_rtcp/source/flexfec_03_header_reader_writer_unittest.cc
@@ -24,6 +24,15 @@
 #include "test/gmock.h"
 #include "test/gtest.h"
 
+#define kMaxMediaPackets          kMaxMediaPackets_Flexfec03HeaderReaderWriter
+#define kMaxTrackedMediaPackets   kMaxTrackedMediaPackets_Flexfec03HeaderReaderWriter
+#define kMaxFecPackets            kMaxFecPackets_Flexfec03HeaderReaderWriter
+#define kFlexfecPacketMaskSizes   kFlexfecPacketMaskSizes_Flexfec03HeaderReaderWriter
+#define kBaseHeaderSize           kBaseHeaderSize_Flexfec03HeaderReaderWriter
+#define kStreamSpecificHeaderSize kStreamSpecificHeaderSize_Flexfec03HeaderReaderWriter
+#define kHeaderSizes              kHeaderSizes_Flexfec03HeaderReaderWriter
+#define FlexfecHeaderSize         FlexfecHeaderSize_Flexfec03HeaderReaderWriter
+
 namespace webrtc {
 
 namespace {
@@ -575,3 +584,12 @@ TEST(Flexfec03HeaderReaderWriterTest,
 }
 
 }  // namespace webrtc
+
+#undef kMaxMediaPackets
+#undef kMaxTrackedMediaPackets
+#undef kMaxFecPackets
+#undef kFlexfecPacketMaskSizes
+#undef kBaseHeaderSize
+#undef kStreamSpecificHeaderSize
+#undef kHeaderSizes
+#undef FlexfecHeaderSize
diff --git a/third_party/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc b/third_party/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc
index d1e0b99d3b..f5480b89b0 100644
--- a/third_party/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc
+++ b/third_party/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc
@@ -22,6 +22,15 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 
+#define kMaxMediaPackets          kMaxMediaPackets_FlexfecHeaderReaderWriter
+#define kMaxTrackedMediaPackets   kMaxTrackedMediaPackets_FlexfecHeaderReaderWriter
+#define kMaxFecPackets            kMaxFecPackets_FlexfecHeaderReaderWriter
+#define kFlexfecPacketMaskSizes   kFlexfecPacketMaskSizes_FlexfecHeaderReaderWriter
+#define kBaseHeaderSize           kBaseHeaderSize_FlexfecHeaderReaderWriter
+#define kStreamSpecificHeaderSize kStreamSpecificHeaderSize_FlexfecHeaderReaderWriter
+#define kHeaderSizes              kHeaderSizes_FlexfecHeaderReaderWriter
+#define FlexfecHeaderSize         FlexfecHeaderSize_FlexfecHeaderReaderWriter
+
 namespace webrtc {
 
 namespace {
@@ -330,3 +339,12 @@ void FlexfecHeaderWriter::FinalizeFecHeader(
 }
 
 }  // namespace webrtc
+
+#undef kMaxMediaPackets
+#undef kMaxTrackedMediaPackets
+#undef kMaxFecPackets
+#undef kFlexfecPacketMaskSizes
+#undef kBaseHeaderSize
+#undef kStreamSpecificHeaderSize
+#undef kHeaderSizes
+#undef FlexfecHeaderSize
diff --git a/third_party/webrtc/modules/rtp_rtcp/source/flexfec_receiver.cc b/third_party/webrtc/modules/rtp_rtcp/source/flexfec_receiver.cc
index c6465fbccf..c6936b9a31 100644
--- a/third_party/webrtc/modules/rtp_rtcp/source/flexfec_receiver.cc
+++ b/third_party/webrtc/modules/rtp_rtcp/source/flexfec_receiver.cc
@@ -27,6 +27,8 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 
+#define kPacketLogInterval kPacketLogInterval_FlexfecReceiver
+
 namespace webrtc {
 
 namespace {
@@ -207,3 +209,5 @@ void FlexfecReceiver::ProcessReceivedPacket(
 }
 
 }  // namespace webrtc
+
+#undef kPacketLogInterval
diff --git a/third_party/webrtc/modules/rtp_rtcp/source/flexfec_sender.cc b/third_party/webrtc/modules/rtp_rtcp/source/flexfec_sender.cc
index 9b5ceac577..53f10604a8 100644
--- a/third_party/webrtc/modules/rtp_rtcp/source/flexfec_sender.cc
+++ b/third_party/webrtc/modules/rtp_rtcp/source/flexfec_sender.cc
@@ -38,6 +38,8 @@
 #include "rtc_base/race_checker.h"
 #include "rtc_base/synchronization/mutex.h"
 
+#define kPacketLogInterval kPacketLogInterval_FlexfecSender
+
 namespace webrtc {
 
 namespace {
@@ -216,3 +218,5 @@ std::optional<RtpState> FlexfecSender::GetRtpState() {
 }
 
 }  // namespace webrtc
+
+#undef kPacketLogInterval
diff --git a/third_party/webrtc/modules/rtp_rtcp/source/packet_sequencer.cc b/third_party/webrtc/modules/rtp_rtcp/source/packet_sequencer.cc
index 930ace9d38..b31eae4d3b 100644
--- a/third_party/webrtc/modules/rtp_rtcp/source/packet_sequencer.cc
+++ b/third_party/webrtc/modules/rtp_rtcp/source/packet_sequencer.cc
@@ -22,6 +22,9 @@
 #include "rtc_base/random.h"
 #include "system_wrappers/include/clock.h"
 
+#define kTimestampTicksPerMs   kTimestampTicksPerMs_PacketSequencer
+#define kRedForFecHeaderLength kRedForFecHeaderLength_PacketSequencer
+
 namespace webrtc {
 
 namespace {
@@ -159,3 +162,6 @@ bool PacketSequencer::CanSendPaddingOnMediaSsrc() const {
 }
 
 }  // namespace webrtc
+
+#undef kTimestampTicksPerMs
+#undef kRedForFecHeaderLength
diff --git a/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index cf9975e764..82ad96c6f0 100644
--- a/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -71,6 +71,9 @@
 #include "system_wrappers/include/clock.h"
 #include "system_wrappers/include/ntp_time.h"
 
+#define kDefaultVideoReportInterval kDefaultVideoReportInterval_RTCPReceiver
+#define kDefaultAudioReportInterval kDefaultAudioReportInterval_RTCPReceiver
+
 namespace webrtc {
 namespace {
 
@@ -1252,3 +1255,6 @@ bool RTCPReceiver::RtcpRrSequenceNumberTimeoutLocked(Timestamp now) {
 }
 
 }  // namespace webrtc
+
+#undef kDefaultVideoReportInterval
+#undef kDefaultAudioReportInterval
diff --git a/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index e06698e0db..f944bc11e1 100644
--- a/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -62,6 +62,9 @@
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/trace_event.h"
 
+#define kDefaultVideoReportInterval kDefaultVideoReportInterval_RTCPSender
+#define kDefaultAudioReportInterval kDefaultAudioReportInterval_RTCPSender
+
 namespace webrtc {
 
 namespace {
@@ -908,3 +911,6 @@ void RTCPSender::SetNextRtcpSendEvaluationDuration(TimeDelta duration) {
 }
 
 }  // namespace webrtc
+
+#undef kDefaultVideoReportInterval
+#undef kDefaultAudioReportInterval
diff --git a/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet.cc b/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
index fe9d3a7d95..445206bfc1 100644
--- a/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
+++ b/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
@@ -27,6 +27,8 @@
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/strings/string_builder.h"
 
+#define kRtpVersion kRtpVersion_RtpPacket
+
 namespace webrtc {
 namespace {
 constexpr size_t kFixedHeaderSize = 12;
@@ -702,3 +704,5 @@ std::string RtpPacket::ToString() const {
 }
 
 }  // namespace webrtc
+
+#undef kRtpVersion
diff --git a/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_egress.cc b/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_egress.cc
index 57758ad538..2b108603d9 100644
--- a/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_egress.cc
+++ b/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_egress.cc
@@ -44,6 +44,8 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/task_utils/repeating_task.h"
 
+#define kTimestampTicksPerMs kTimestampTicksPerMs_RtpSenderEgress
+
 namespace webrtc {
 namespace {
 constexpr uint32_t kTimestampTicksPerMs = 90;
@@ -515,3 +517,5 @@ void RtpSenderEgress::PeriodicUpdate() {
       send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_);
 }
 }  // namespace webrtc
+
+#undef kTimestampTicksPerMs
diff --git a/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index 8503b0db75..8806b6779d 100644
--- a/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -69,6 +69,8 @@
 #include "rtc_base/synchronization/mutex.h"
 #include "system_wrappers/include/ntp_time.h"
 
+#define kRedForFecHeaderLength kRedForFecHeaderLength_RtpSenderVideo
+
 namespace webrtc {
 
 namespace {
@@ -924,3 +926,5 @@ void RTPSenderVideo::MaybeUpdateCurrentPlayoutDelay(
 }
 
 }  // namespace webrtc
+
+#undef kRedForFecHeaderLength
diff --git a/third_party/webrtc/modules/rtp_rtcp/source/rtp_util.cc b/third_party/webrtc/modules/rtp_rtcp/source/rtp_util.cc
index 4d802b6308..310542f1ef 100644
--- a/third_party/webrtc/modules/rtp_rtcp/source/rtp_util.cc
+++ b/third_party/webrtc/modules/rtp_rtcp/source/rtp_util.cc
@@ -17,6 +17,8 @@
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "rtc_base/checks.h"
 
+#define kRtpVersion kRtpVersion_RtpUtil
+
 namespace webrtc {
 namespace {
 
@@ -61,3 +63,5 @@ uint32_t ParseRtpSsrc(ArrayView<const uint8_t> rtp_packet) {
 }
 
 }  // namespace webrtc
+
+#undef kRtpVersion
diff --git a/third_party/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc b/third_party/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
index 885d44bdf0..0dfbeafe94 100644
--- a/third_party/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
+++ b/third_party/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
@@ -35,12 +35,17 @@ namespace webrtc {
 namespace {
 using BufferFlags = Vp8FrameConfig::BufferFlags;
 
-constexpr BufferFlags kNone = Vp8FrameConfig::BufferFlags::kNone;
-constexpr BufferFlags kReference = Vp8FrameConfig::BufferFlags::kReference;
-constexpr BufferFlags kUpdate = Vp8FrameConfig::BufferFlags::kUpdate;
-constexpr BufferFlags kReferenceAndUpdate =
+constexpr BufferFlags kNone_ScreenshareLayers = Vp8FrameConfig::BufferFlags::kNone;
+constexpr BufferFlags kReference_ScreenshareLayers = Vp8FrameConfig::BufferFlags::kReference;
+constexpr BufferFlags kUpdate_ScreenshareLayers = Vp8FrameConfig::BufferFlags::kUpdate;
+constexpr BufferFlags kReferenceAndUpdate_ScreenshareLayers =
     Vp8FrameConfig::BufferFlags::kReferenceAndUpdate;
 
+#define kNone               kNone_ScreenshareLayers
+#define kReference          kReference_ScreenshareLayers
+#define kUpdate             kUpdate_ScreenshareLayers
+#define kReferenceAndUpdate kUpdate_ScreenshareLayers
+
 constexpr int kOneSecond90Khz = 90000;
 constexpr int kMinTimeBetweenSyncs = kOneSecond90Khz * 2;
 constexpr int kMaxTimeBetweenSyncs = kOneSecond90Khz * 4;
@@ -631,3 +636,8 @@ void ScreenshareLayers::UpdateHistograms() {
   }
 }
 }  // namespace webrtc
+
+#undef kNone
+#undef kReference
+#undef kUpdate
+#undef kReferenceAndUpdate
diff --git a/third_party/webrtc/modules/video_coding/svc/scalability_structure_l2t2_key_shift.cc b/third_party/webrtc/modules/video_coding/svc/scalability_structure_l2t2_key_shift.cc
index b39b953767..f2510c5769 100644
--- a/third_party/webrtc/modules/video_coding/svc/scalability_structure_l2t2_key_shift.cc
+++ b/third_party/webrtc/modules/video_coding/svc/scalability_structure_l2t2_key_shift.cc
@@ -17,6 +17,8 @@
 #include "modules/video_coding/svc/scalable_video_controller.h"
 #include "rtc_base/checks.h"
 
+#define Dti Dti_ScalabilityStructureL2T2KeyShift
+
 namespace webrtc {
 namespace {
 
@@ -175,3 +177,5 @@ void ScalabilityStructureL2T2KeyShift::OnRatesUpdated(
 }
 
 }  // namespace webrtc
+
+#undef Dti
diff --git a/third_party/webrtc/modules/video_coding/svc/scalability_structure_simulcast.cc b/third_party/webrtc/modules/video_coding/svc/scalability_structure_simulcast.cc
index 71eae32794..7d39c7f2ee 100644
--- a/third_party/webrtc/modules/video_coding/svc/scalability_structure_simulcast.cc
+++ b/third_party/webrtc/modules/video_coding/svc/scalability_structure_simulcast.cc
@@ -18,6 +18,8 @@
 #include "modules/video_coding/svc/scalable_video_controller.h"
 #include "rtc_base/checks.h"
 
+#define Dti Dti_ScalabilityStructureSimulcast
+
 namespace webrtc {
 namespace {
 
@@ -352,3 +354,5 @@ FrameDependencyStructure ScalabilityStructureS3T3::DependencyStructure() const {
 }
 
 }  // namespace webrtc
+
+#undef Dti
diff --git a/third_party/webrtc/pc/jsep_session_description.cc b/third_party/webrtc/pc/jsep_session_description.cc
index bbeb4c3c07..fb57d6e32c 100644
--- a/third_party/webrtc/pc/jsep_session_description.cc
+++ b/third_party/webrtc/pc/jsep_session_description.cc
@@ -36,6 +36,9 @@
 using webrtc::Candidate;
 using ::webrtc::SessionDescription;
 
+#define kDummyAddress kDummyAddress_JsepSessionDescription
+#define kDummyPort    kDummyPort_JsepSessionDescription
+
 namespace webrtc {
 namespace {
 
@@ -364,3 +367,6 @@ int JsepSessionDescription::GetMediasectionIndex(const Candidate& candidate) {
 }
 
 }  // namespace webrtc
+
+#undef kDummyAddress
+#undef kDummyPort
diff --git a/third_party/webrtc/pc/webrtc_sdp.cc b/third_party/webrtc/pc/webrtc_sdp.cc
index 6e434676f9..464aa79184 100644
--- a/third_party/webrtc/pc/webrtc_sdp.cc
+++ b/third_party/webrtc/pc/webrtc_sdp.cc
@@ -69,6 +69,9 @@
 #include "rtc_base/string_encode.h"
 #include "rtc_base/strings/string_builder.h"
 
+#define kDummyAddress kDummyAddress_WebRTCSDP
+#define kDummyPort    kDummyPort_WebRTCSDP
+
 // TODO(deadbeef): Switch to using anonymous namespace rather than declaring
 // everything "static".
 namespace webrtc {
@@ -3761,3 +3764,6 @@
 }
 
 }  // namespace webrtc
+
+#undef kDummyAddress
+#undef kDummyPort
diff --git a/third_party/webrtc/webrtc.gni b/third_party/webrtc/webrtc.gni
index 103983dec4..58bbffea9c 100644
--- a/third_party/webrtc/webrtc.gni
+++ b/third_party/webrtc/webrtc.gni
@@ -7,6 +7,7 @@
 # be found in the AUTHORS file in the root of the source tree.
 import("//build/config/arm.gni")
 import("//build/config/features.gni")
+import("//build/config/jumbo.gni")
 import("//build/config/mips.gni")
 import("//build/config/ozone.gni")
 import("//build/config/sanitizers/sanitizers.gni")
@@ -587,7 +588,7 @@ template("rtc_test") {
 }
 
 template("rtc_source_set") {
-  source_set(target_name) {
+  jumbo_source_set(target_name) {
     forward_variables_from(invoker,
                            "*",
                            [
@@ -700,7 +701,7 @@ template("rtc_source_set") {
 }
 
 template("rtc_static_library") {
-  static_library(target_name) {
+  jumbo_static_library(target_name) {
     forward_variables_from(invoker,
                            "*",
                            [
@@ -823,9 +824,9 @@ template("rtc_library") {
   # source files will cause issues with macOS libtool.
   if (header_only || is_component_build ||
       (defined(invoker.testonly) && invoker.testonly)) {
-    target_type = "source_set"
+    target_type = "jumbo_source_set"
   } else {
-    target_type = "static_library"
+    target_type = "jumbo_static_library"
   }
   target(target_type, target_name) {
     forward_variables_from(invoker,
diff --git a/third_party/webrtc/modules/audio_processing/aec3/fullband_erle_estimator.cc b/third_party/webrtc/modules/audio_processing/aec3/fullband_erle_estimator.cc
index 498c21b568..b375d805ba 100644
--- a/third_party/webrtc/modules/audio_processing/aec3/fullband_erle_estimator.cc
+++ b/third_party/webrtc/modules/audio_processing/aec3/fullband_erle_estimator.cc
@@ -21,6 +21,10 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_minmax.h"
 
+#define kX2BandEnergyThreshold kX2BandEnergyThreshold_FullBandErleEstimator
+#define kPointsToAccumulate    kPointsToAccumulate_FullBandErleEstimator
+#define kBlocksToHoldErle      kBlocksToHoldErle_FullBandErleEstimator
+
 namespace webrtc {
 
 namespace {
@@ -189,3 +193,7 @@ void FullBandErleEstimator::ErleInstantaneous::UpdateQualityEstimate() {
 }
 
 }  // namespace webrtc
+
+#undef kX2BandEnergyThreshold
+#undef kPointsToAccumulate
+#undef kBlocksToHoldErle
diff --git a/third_party/webrtc/modules/audio_processing/aec3/subband_erle_estimator.cc b/third_party/webrtc/modules/audio_processing/aec3/subband_erle_estimator.cc
index 6208d26388..906e9a4dfc 100644
--- a/third_party/webrtc/modules/audio_processing/aec3/subband_erle_estimator.cc
+++ b/third_party/webrtc/modules/audio_processing/aec3/subband_erle_estimator.cc
@@ -18,6 +18,10 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_minmax.h"
 
+#define kX2BandEnergyThreshold kX2BandEnergyThreshold_SubbandErleEstimator
+#define kPointsToAccumulate    kPointsToAccumulate_SubbandErleEstimator
+#define kBlocksToHoldErle      kBlocksToHoldErle_SubbandErleEstimator
+
 namespace webrtc {
 
 namespace {
@@ -251,3 +255,7 @@ void SubbandErleEstimator::UpdateAccumulatedSpectra(
 }
 
 }  // namespace webrtc
+
+#undef kX2BandEnergyThreshold
+#undef kPointsToAccumulate
+#undef kBlocksToHoldErle
diff --git a/third_party/webrtc/modules/rtp_rtcp/source/rtp_packetizer_av1.cc b/third_party/webrtc/modules/rtp_rtcp/source/rtp_packetizer_av1.cc
index e5293a90e2..3bd33d102a 100644
--- a/third_party/webrtc/modules/rtp_rtcp/source/rtp_packetizer_av1.cc
+++ b/third_party/webrtc/modules/rtp_rtcp/source/rtp_packetizer_av1.cc
@@ -25,6 +25,10 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 
+#define ObuHasExtension    ObuHasExtension_RTPPacketizerAV1
+#define ObuHasSize         ObuHasSize_RTPPacketizerAV1
+#define kObuSizePresentBit kObuSizePresentBit_RTPPacketizerAV1
+
 namespace webrtc {
 namespace {
 constexpr int kAggregationHeaderSize = 1;
@@ -456,3 +460,7 @@ bool RtpPacketizerAv1::NextPacket(RtpPacketToSend* packet) {
 }
 
 }  // namespace webrtc
+
+#undef ObuHasExtension
+#undef ObuHasSize
+#undef kObuSizePresentBit
diff --git a/third_party/webrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc b/third_party/webrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
index fd8d647045..3695dad955 100644
--- a/third_party/webrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
+++ b/third_party/webrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
@@ -34,6 +34,10 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_conversions.h"
 
+#define ObuHasExtension    ObuHasExtension_VideoRTPDepacketizerAV1
+#define ObuHasSize         ObuHasSize_VideoRTPDepacketizerAV1
+#define kObuSizePresentBit kObuSizePresentBit_VideoRTPDepacketizerAV1
+
 namespace webrtc {
 namespace {
 // AV1 format:
@@ -404,3 +408,7 @@ VideoRtpDepacketizerAv1::Parse(CopyOnWriteBuffer rtp_payload) {
 }
 
 }  // namespace webrtc
+
+#undef ObuHasExtension
+#undef ObuHasSize
+#undef kObuSizePresentBit