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[cleanup] Migrate absl::in_place to std::in_place

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Self-explanatory.

Fixed: webrtc:342905193
Change-Id: I3cf1ec99ef6867bb33289977246725badf2bfcfe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363360
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}
This commit is contained in:
Ho Cheung 2024-09-23 21:58:08 +08:00 committed by WebRTC LUCI CQ
parent 5a294aeea3
commit a8efbb223b
8 changed files with 14 additions and 11 deletions
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3
modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
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@ -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.

2
modules/rtp_rtcp/source/video_rtp_depacketizer_generic.cc
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@ -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];

4
modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc
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@ -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;
@ -234,7 +234,7 @@ 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;

4
modules/rtp_rtcp/source/video_rtp_depacketizer_h265.cc
View File

@ -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;

2
modules/rtp_rtcp/source/video_rtp_depacketizer_raw.cc
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@ -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;
}

2
modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc
View File

@ -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;

4
modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc
View File

@ -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;

4
rtc_base/bitstream_reader_unittest.cc
View File

@ -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)