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termux-packages/x11-packages/chromium-host-tools/jumbo-patches/0115-webrtc.patch
Chongyun Lee dda7b183ae chromium-host-tools: bump to 139.0.7258.154 (#25872)
2025-08-31 05:18:44 +08:00

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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
@@ -67,6 +67,9 @@
#include "rtc_base/string_encode.h"
#include "rtc_base/strings/string_builder.h"
+#define kDummyAddress kDummyAddress_WebRTCRDP
+#define kDummyPort kDummyPort_WebRTCRDP
+
namespace webrtc {
namespace {
@@ -3619,3 +3622,6 @@ bool ParseFmtpParameterSet(absl::string_view line_params,
}
} // 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,
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