mozilla-central/third_party/libwebrtc/moz-patch-stack/0133.patch

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From: Michael Froman <mjfroman@mac.com>
Date: Mon, 14 Oct 2024 17:39:14 -0500
Subject: Bug 1924098 - Cherry-pick upstream libwebrtc commit a8efbb223b r?ng
[cleanup] Migrate `absl::in_place` to `std::in_place`
Self-explanatory.
Fixed: webrtc:342905193
Change-Id: I3cf1ec99ef6867bb33289977246725badf2bfcfe
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Ho Cheung <hocheung@chromium.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43071}
---
modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc | 3 ++-
modules/rtp_rtcp/source/video_rtp_depacketizer_generic.cc | 2 +-
modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc | 4 ++--
modules/rtp_rtcp/source/video_rtp_depacketizer_h265.cc | 4 ++--
modules/rtp_rtcp/source/video_rtp_depacketizer_raw.cc | 2 +-
modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc | 2 +-
modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc | 4 +++-
rtc_base/bitstream_reader_unittest.cc | 4 ++--
8 files changed, 14 insertions(+), 11 deletions(-)
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
index 6f70297447..16dcbb59ab 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
@@ -13,6 +13,7 @@
#include <stddef.h>
#include <stdint.h>
+#include <optional>
#include <utility>
#include "modules/rtp_rtcp/source/leb128.h"
@@ -367,7 +368,7 @@ VideoRtpDepacketizerAv1::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
// new coded video sequence can't start from an OBU fragment.
return std::nullopt;
}
- std::optional<ParsedRtpPayload> parsed(absl::in_place);
+ std::optional<ParsedRtpPayload> parsed(std::in_place);
// To assemble frame, all of the rtp payload is required, including
// aggregation header.
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.cc
index ec8993b191..344bb3f1e1 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.cc
@@ -39,7 +39,7 @@ VideoRtpDepacketizerGeneric::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
RTC_LOG(LS_WARNING) << "Empty payload.";
return std::nullopt;
}
- std::optional<ParsedRtpPayload> parsed(absl::in_place);
+ std::optional<ParsedRtpPayload> parsed(std::in_place);
const uint8_t* payload_data = rtp_payload.cdata();
uint8_t generic_header = payload_data[0];
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc
index e8fd911efe..16d1c2c14c 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc
@@ -63,7 +63,7 @@ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> ProcessStapAOrSingleNalu(
rtc::CopyOnWriteBuffer rtp_payload) {
const uint8_t* const payload_data = rtp_payload.cdata();
std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload(
- absl::in_place);
+ std::in_place);
bool modified_buffer = false;
parsed_payload->video_payload = rtp_payload;
parsed_payload->video_header.width = 0;
@@ -235,7 +235,7 @@ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> ParseFuaNalu(
return std::nullopt;
}
std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload(
- absl::in_place);
+ std::in_place);
uint8_t fnri = rtp_payload.cdata()[0] & (kH264FBit | kH264NriMask);
uint8_t original_nal_type = rtp_payload.cdata()[1] & kH264TypeMask;
bool first_fragment = (rtp_payload.cdata()[1] & kH264SBit) > 0;
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.cc
index d16e638d06..e0ed2215af 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_h265.cc
@@ -71,7 +71,7 @@ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> ProcessApOrSingleNalu(
}
const uint8_t* const payload_data = rtp_payload.cdata();
std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload(
- absl::in_place);
+ std::in_place);
parsed_payload->video_header.width = 0;
parsed_payload->video_header.height = 0;
parsed_payload->video_header.codec = kVideoCodecH265;
@@ -191,7 +191,7 @@ std::optional<VideoRtpDepacketizer::ParsedRtpPayload> ParseFuNalu(
return std::nullopt;
}
std::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload(
- absl::in_place);
+ std::in_place);
uint8_t f = rtp_payload.cdata()[0] & kH265FBit;
uint8_t layer_id_h = rtp_payload.cdata()[0] & kH265LayerIDHMask;
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.cc
index f4f4a794ea..f18e344baa 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.cc
@@ -20,7 +20,7 @@ namespace webrtc {
std::optional<VideoRtpDepacketizer::ParsedRtpPayload>
VideoRtpDepacketizerRaw::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
- std::optional<ParsedRtpPayload> parsed(absl::in_place);
+ std::optional<ParsedRtpPayload> parsed(std::in_place);
parsed->video_payload = std::move(rtp_payload);
return parsed;
}
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc
index 157ad37aaa..709415771f 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc
@@ -136,7 +136,7 @@ std::optional<VideoRtpDepacketizer::ParsedRtpPayload>
VideoRtpDepacketizerVp8::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
rtc::ArrayView<const uint8_t> payload(rtp_payload.cdata(),
rtp_payload.size());
- std::optional<ParsedRtpPayload> result(absl::in_place);
+ std::optional<ParsedRtpPayload> result(std::in_place);
int offset = ParseRtpPayload(payload, &result->video_header);
if (offset == kFailedToParse)
return std::nullopt;
diff --git a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc
index ebd27ffdea..574c7c8530 100644
--- a/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc
+++ b/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc
@@ -12,6 +12,8 @@
#include <string.h>
+#include <optional>
+
#include "api/video/video_codec_constants.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "modules/video_coding/codecs/interface/common_constants.h"
@@ -148,7 +150,7 @@ void ParseSsData(BitstreamReader& parser, RTPVideoHeaderVP9* vp9) {
std::optional<VideoRtpDepacketizer::ParsedRtpPayload>
VideoRtpDepacketizerVp9::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
- std::optional<ParsedRtpPayload> result(absl::in_place);
+ std::optional<ParsedRtpPayload> result(std::in_place);
int offset = ParseRtpPayload(rtp_payload, &result->video_header);
if (offset == 0)
return std::nullopt;
diff --git a/rtc_base/bitstream_reader_unittest.cc b/rtc_base/bitstream_reader_unittest.cc
index b9b51a2571..3a48e7d086 100644
--- a/rtc_base/bitstream_reader_unittest.cc
+++ b/rtc_base/bitstream_reader_unittest.cc
@@ -28,7 +28,7 @@ namespace {
TEST(BitstreamReaderTest, InDebugModeRequiresToCheckOkStatusBeforeDestruction) {
const uint8_t bytes[32] = {};
- std::optional<BitstreamReader> reader(absl::in_place, bytes);
+ std::optional<BitstreamReader> reader(std::in_place, bytes);
EXPECT_GE(reader->ReadBits(7), 0u);
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(OS_ANDROID)
@@ -40,7 +40,7 @@ TEST(BitstreamReaderTest, InDebugModeRequiresToCheckOkStatusBeforeDestruction) {
TEST(BitstreamReaderTest, InDebugModeMayCheckRemainingBitsInsteadOfOkStatus) {
const uint8_t bytes[32] = {};
- std::optional<BitstreamReader> reader(absl::in_place, bytes);
+ std::optional<BitstreamReader> reader(std::in_place, bytes);
EXPECT_GE(reader->ReadBit(), 0);
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(OS_ANDROID)