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Replace rtc::Optional with absl::optional

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This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script passing top level directories except rtc_base and api

find $@ -type f \( -name \*.h -o -name \*.cc -o -name \*.mm \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: I9465c172e65ba6e6ed4e4fdc35b0b265038d6f71
Reviewed-on: https://webrtc-review.googlesource.com/84584
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23697}
This commit is contained in:
Danil Chapovalov 2018-06-21 10:17:24 +02:00 committed by Commit Bot
parent ae810c10b4
commit 196100efa6
71 changed files with 189 additions and 188 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
common_audio/BUILD.gn
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@ -64,7 +64,6 @@ rtc_static_library("common_audio") {
":sinc_resampler", ":sinc_resampler",
"..:webrtc_common", "..:webrtc_common",
"../:typedefs", "../:typedefs",
"../api:optional",
"../rtc_base:checks", "../rtc_base:checks",
"../rtc_base:gtest_prod", "../rtc_base:gtest_prod",
"../rtc_base:rtc_base_approved", "../rtc_base:rtc_base_approved",
@ -72,6 +71,7 @@ rtc_static_library("common_audio") {
"../rtc_base/memory:aligned_malloc", "../rtc_base/memory:aligned_malloc",
"../system_wrappers", "../system_wrappers",
"../system_wrappers:cpu_features_api", "../system_wrappers:cpu_features_api",
"//third_party/abseil-cpp/absl/types:optional",
] ]
defines = [] defines = []

2
common_audio/mocks/mock_smoothing_filter.h
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@ -19,7 +19,7 @@ namespace webrtc {
class MockSmoothingFilter : public SmoothingFilter { class MockSmoothingFilter : public SmoothingFilter {
public: public:
MOCK_METHOD1(AddSample, void(float)); MOCK_METHOD1(AddSample, void(float));
MOCK_METHOD0(GetAverage, rtc::Optional<float>()); MOCK_METHOD0(GetAverage, absl::optional<float>());
MOCK_METHOD1(SetTimeConstantMs, bool(int)); MOCK_METHOD1(SetTimeConstantMs, bool(int));
}; };

4
common_audio/smoothing_filter.cc
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@ -52,10 +52,10 @@ void SmoothingFilterImpl::AddSample(float sample) {
last_sample_ = sample; last_sample_ = sample;
} }
rtc::Optional<float> SmoothingFilterImpl::GetAverage() { absl::optional<float> SmoothingFilterImpl::GetAverage() {
if (!init_end_time_ms_) { if (!init_end_time_ms_) {
// |init_end_time_ms_| undefined since we have not received any sample. // |init_end_time_ms_| undefined since we have not received any sample.
return rtc::nullopt; return absl::nullopt;
} }
ExtrapolateLastSample(rtc::TimeMillis()); ExtrapolateLastSample(rtc::TimeMillis());
return state_; return state_;

8
common_audio/smoothing_filter.h
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@ -11,7 +11,7 @@
#ifndef COMMON_AUDIO_SMOOTHING_FILTER_H_ #ifndef COMMON_AUDIO_SMOOTHING_FILTER_H_
#define COMMON_AUDIO_SMOOTHING_FILTER_H_ #define COMMON_AUDIO_SMOOTHING_FILTER_H_
#include "api/optional.h" #include "absl/types/optional.h"
#include "rtc_base/constructormagic.h" #include "rtc_base/constructormagic.h"
#include "system_wrappers/include/clock.h" #include "system_wrappers/include/clock.h"
@ -21,7 +21,7 @@ class SmoothingFilter {
public: public:
virtual ~SmoothingFilter() = default; virtual ~SmoothingFilter() = default;
virtual void AddSample(float sample) = 0; virtual void AddSample(float sample) = 0;
virtual rtc::Optional<float> GetAverage() = 0; virtual absl::optional<float> GetAverage() = 0;
virtual bool SetTimeConstantMs(int time_constant_ms) = 0; virtual bool SetTimeConstantMs(int time_constant_ms) = 0;
}; };
@ -44,7 +44,7 @@ class SmoothingFilterImpl final : public SmoothingFilter {
~SmoothingFilterImpl() override; ~SmoothingFilterImpl() override;
void AddSample(float sample) override; void AddSample(float sample) override;
rtc::Optional<float> GetAverage() override; absl::optional<float> GetAverage() override;
bool SetTimeConstantMs(int time_constant_ms) override; bool SetTimeConstantMs(int time_constant_ms) override;
// Methods used for unittests. // Methods used for unittests.
@ -58,7 +58,7 @@ class SmoothingFilterImpl final : public SmoothingFilter {
const float init_factor_; const float init_factor_;
const float init_const_; const float init_const_;
rtc::Optional<int64_t> init_end_time_ms_; absl::optional<int64_t> init_end_time_ms_;
float last_sample_; float last_sample_;
float alpha_; float alpha_;
float state_; float state_;

2
common_video/BUILD.gn
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@ -57,7 +57,6 @@ rtc_static_library("common_video") {
deps = [ deps = [
"..:webrtc_common", "..:webrtc_common",
"../:typedefs", "../:typedefs",
"../api:optional",
"../api/video:video_bitrate_allocation", "../api/video:video_bitrate_allocation",
"../api/video:video_frame", "../api/video:video_frame",
"../api/video:video_frame_i420", "../api/video:video_frame_i420",
@ -67,6 +66,7 @@ rtc_static_library("common_video") {
"../rtc_base:rtc_base", "../rtc_base:rtc_base",
"../rtc_base:rtc_task_queue", "../rtc_base:rtc_task_queue",
"../rtc_base:safe_minmax", "../rtc_base:safe_minmax",
"//third_party/abseil-cpp/absl/types:optional",
"//third_party/libyuv", "//third_party/libyuv",
] ]
} }

2
common_video/bitrate_adjuster.cc
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@ -68,7 +68,7 @@ uint32_t BitrateAdjuster::GetAdjustedBitrateBps() const {
return adjusted_bitrate_bps_; return adjusted_bitrate_bps_;
} }
rtc::Optional<uint32_t> BitrateAdjuster::GetEstimatedBitrateBps() { absl::optional<uint32_t> BitrateAdjuster::GetEstimatedBitrateBps() {
rtc::CritScope cs(&crit_); rtc::CritScope cs(&crit_);
return bitrate_tracker_.Rate(rtc::TimeMillis()); return bitrate_tracker_.Rate(rtc::TimeMillis());
} }

2
common_video/h264/h264_bitstream_parser.cc
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@ -44,7 +44,7 @@ H264BitstreamParser::Result H264BitstreamParser::ParseNonParameterSetNalu(
if (!sps_ || !pps_) if (!sps_ || !pps_)
return kInvalidStream; return kInvalidStream;
last_slice_qp_delta_ = rtc::nullopt; last_slice_qp_delta_ = absl::nullopt;
const std::vector<uint8_t> slice_rbsp = const std::vector<uint8_t> slice_rbsp =
H264::ParseRbsp(source, source_length); H264::ParseRbsp(source, source_length);
if (slice_rbsp.size() < H264::kNaluTypeSize) if (slice_rbsp.size() < H264::kNaluTypeSize)

8
common_video/h264/h264_bitstream_parser.h
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@ -13,7 +13,7 @@
#include <stddef.h> #include <stddef.h>
#include <stdint.h> #include <stdint.h>
#include "api/optional.h" #include "absl/types/optional.h"
#include "common_video/h264/pps_parser.h" #include "common_video/h264/pps_parser.h"
#include "common_video/h264/sps_parser.h" #include "common_video/h264/sps_parser.h"
@ -53,11 +53,11 @@ class H264BitstreamParser {
uint8_t nalu_type); uint8_t nalu_type);
// SPS/PPS state, updated when parsing new SPS/PPS, used to parse slices. // SPS/PPS state, updated when parsing new SPS/PPS, used to parse slices.
rtc::Optional<SpsParser::SpsState> sps_; absl::optional<SpsParser::SpsState> sps_;
rtc::Optional<PpsParser::PpsState> pps_; absl::optional<PpsParser::PpsState> pps_;
// Last parsed slice QP. // Last parsed slice QP.
rtc::Optional<int32_t> last_slice_qp_delta_; absl::optional<int32_t> last_slice_qp_delta_;
}; };
} // namespace webrtc } // namespace webrtc

14
common_video/h264/pps_parser.cc
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@ -19,7 +19,7 @@
#define RETURN_EMPTY_ON_FAIL(x) \ #define RETURN_EMPTY_ON_FAIL(x) \
if (!(x)) { \ if (!(x)) { \
return rtc::nullopt; \ return absl::nullopt; \
} }
namespace { namespace {
@ -33,7 +33,7 @@ namespace webrtc {
// You can find it on this page: // You can find it on this page:
// http://www.itu.int/rec/T-REC-H.264 // http://www.itu.int/rec/T-REC-H.264
rtc::Optional<PpsParser::PpsState> PpsParser::ParsePps(const uint8_t* data, absl::optional<PpsParser::PpsState> PpsParser::ParsePps(const uint8_t* data,
size_t length) { size_t length) {
// First, parse out rbsp, which is basically the source buffer minus emulation // First, parse out rbsp, which is basically the source buffer minus emulation
// bytes (the last byte of a 0x00 0x00 0x03 sequence). RBSP is defined in // bytes (the last byte of a 0x00 0x00 0x03 sequence). RBSP is defined in
@ -57,7 +57,7 @@ bool PpsParser::ParsePpsIds(const uint8_t* data,
return ParsePpsIdsInternal(&bit_buffer, pps_id, sps_id); return ParsePpsIdsInternal(&bit_buffer, pps_id, sps_id);
} }
rtc::Optional<uint32_t> PpsParser::ParsePpsIdFromSlice(const uint8_t* data, absl::optional<uint32_t> PpsParser::ParsePpsIdFromSlice(const uint8_t* data,
size_t length) { size_t length) {
std::vector<uint8_t> unpacked_buffer = H264::ParseRbsp(data, length); std::vector<uint8_t> unpacked_buffer = H264::ParseRbsp(data, length);
rtc::BitBuffer slice_reader(unpacked_buffer.data(), unpacked_buffer.size()); rtc::BitBuffer slice_reader(unpacked_buffer.data(), unpacked_buffer.size());
@ -65,18 +65,18 @@ rtc::Optional<uint32_t> PpsParser::ParsePpsIdFromSlice(const uint8_t* data,
uint32_t golomb_tmp; uint32_t golomb_tmp;
// first_mb_in_slice: ue(v) // first_mb_in_slice: ue(v)
if (!slice_reader.ReadExponentialGolomb(&golomb_tmp)) if (!slice_reader.ReadExponentialGolomb(&golomb_tmp))
return rtc::nullopt; return absl::nullopt;
// slice_type: ue(v) // slice_type: ue(v)
if (!slice_reader.ReadExponentialGolomb(&golomb_tmp)) if (!slice_reader.ReadExponentialGolomb(&golomb_tmp))
return rtc::nullopt; return absl::nullopt;
// pic_parameter_set_id: ue(v) // pic_parameter_set_id: ue(v)
uint32_t slice_pps_id; uint32_t slice_pps_id;
if (!slice_reader.ReadExponentialGolomb(&slice_pps_id)) if (!slice_reader.ReadExponentialGolomb(&slice_pps_id))
return rtc::nullopt; return absl::nullopt;
return slice_pps_id; return slice_pps_id;
} }
rtc::Optional<PpsParser::PpsState> PpsParser::ParseInternal( absl::optional<PpsParser::PpsState> PpsParser::ParseInternal(
rtc::BitBuffer* bit_buffer) { rtc::BitBuffer* bit_buffer) {
PpsState pps; PpsState pps;

8
common_video/h264/pps_parser.h
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@ -11,7 +11,7 @@
#ifndef COMMON_VIDEO_H264_PPS_PARSER_H_ #ifndef COMMON_VIDEO_H264_PPS_PARSER_H_
#define COMMON_VIDEO_H264_PPS_PARSER_H_ #define COMMON_VIDEO_H264_PPS_PARSER_H_
#include "api/optional.h" #include "absl/types/optional.h"
namespace rtc { namespace rtc {
class BitBuffer; class BitBuffer;
@ -38,20 +38,20 @@ class PpsParser {
}; };
// Unpack RBSP and parse PPS state from the supplied buffer. // Unpack RBSP and parse PPS state from the supplied buffer.
static rtc::Optional<PpsState> ParsePps(const uint8_t* data, size_t length); static absl::optional<PpsState> ParsePps(const uint8_t* data, size_t length);
static bool ParsePpsIds(const uint8_t* data, static bool ParsePpsIds(const uint8_t* data,
size_t length, size_t length,
uint32_t* pps_id, uint32_t* pps_id,
uint32_t* sps_id); uint32_t* sps_id);
static rtc::Optional<uint32_t> ParsePpsIdFromSlice(const uint8_t* data, static absl::optional<uint32_t> ParsePpsIdFromSlice(const uint8_t* data,
size_t length); size_t length);
protected: protected:
// Parse the PPS state, for a bit buffer where RBSP decoding has already been // Parse the PPS state, for a bit buffer where RBSP decoding has already been
// performed. // performed.
static rtc::Optional<PpsState> ParseInternal(rtc::BitBuffer* bit_buffer); static absl::optional<PpsState> ParseInternal(rtc::BitBuffer* bit_buffer);
static bool ParsePpsIdsInternal(rtc::BitBuffer* bit_buffer, static bool ParsePpsIdsInternal(rtc::BitBuffer* bit_buffer,
uint32_t* pps_id, uint32_t* pps_id,
uint32_t* sps_id); uint32_t* sps_id);

4
common_video/h264/pps_parser_unittest.cc
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@ -192,7 +192,7 @@ class PpsParserTest : public ::testing::Test {
PpsParser::PpsState generated_pps_; PpsParser::PpsState generated_pps_;
rtc::Buffer buffer_; rtc::Buffer buffer_;
rtc::Optional<PpsParser::PpsState> parsed_pps_; absl::optional<PpsParser::PpsState> parsed_pps_;
}; };
TEST_F(PpsParserTest, ZeroPps) { TEST_F(PpsParserTest, ZeroPps) {
@ -215,7 +215,7 @@ TEST_F(PpsParserTest, MaxPps) {
} }
TEST_F(PpsParserTest, PpsIdFromSlice) { TEST_F(PpsParserTest, PpsIdFromSlice) {
rtc::Optional<uint32_t> pps_id = PpsParser::ParsePpsIdFromSlice( absl::optional<uint32_t> pps_id = PpsParser::ParsePpsIdFromSlice(
kH264BitstreamChunk, sizeof(kH264BitstreamChunk)); kH264BitstreamChunk, sizeof(kH264BitstreamChunk));
ASSERT_TRUE(pps_id); ASSERT_TRUE(pps_id);
EXPECT_EQ(2u, *pps_id); EXPECT_EQ(2u, *pps_id);

4
common_video/h264/profile_level_id_unittest.cc
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@ -125,7 +125,7 @@ TEST(H264ProfileLevelId, TestToStringInvalid) {
} }
TEST(H264ProfileLevelId, TestParseSdpProfileLevelIdEmpty) { TEST(H264ProfileLevelId, TestParseSdpProfileLevelIdEmpty) {
const rtc::Optional<ProfileLevelId> profile_level_id = const absl::optional<ProfileLevelId> profile_level_id =
ParseSdpProfileLevelId(CodecParameterMap()); ParseSdpProfileLevelId(CodecParameterMap());
EXPECT_TRUE(profile_level_id); EXPECT_TRUE(profile_level_id);
EXPECT_EQ(kProfileConstrainedBaseline, profile_level_id->profile); EXPECT_EQ(kProfileConstrainedBaseline, profile_level_id->profile);
@ -135,7 +135,7 @@ TEST(H264ProfileLevelId, TestParseSdpProfileLevelIdEmpty) {
TEST(H264ProfileLevelId, TestParseSdpProfileLevelIdConstrainedHigh) { TEST(H264ProfileLevelId, TestParseSdpProfileLevelIdConstrainedHigh) {
CodecParameterMap params; CodecParameterMap params;
params["profile-level-id"] = "640c2a"; params["profile-level-id"] = "640c2a";
const rtc::Optional<ProfileLevelId> profile_level_id = const absl::optional<ProfileLevelId> profile_level_id =
ParseSdpProfileLevelId(params); ParseSdpProfileLevelId(params);
EXPECT_TRUE(profile_level_id); EXPECT_TRUE(profile_level_id);
EXPECT_EQ(kProfileConstrainedHigh, profile_level_id->profile); EXPECT_EQ(kProfileConstrainedHigh, profile_level_id->profile);

6
common_video/h264/sps_parser.cc
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@ -18,7 +18,7 @@
#include "rtc_base/logging.h" #include "rtc_base/logging.h"
namespace { namespace {
typedef rtc::Optional<webrtc::SpsParser::SpsState> OptionalSps; typedef absl::optional<webrtc::SpsParser::SpsState> OptionalSps;
#define RETURN_EMPTY_ON_FAIL(x) \ #define RETURN_EMPTY_ON_FAIL(x) \
if (!(x)) { \ if (!(x)) { \
@ -38,14 +38,14 @@ SpsParser::SpsState::SpsState() = default;
// http://www.itu.int/rec/T-REC-H.264 // http://www.itu.int/rec/T-REC-H.264
// Unpack RBSP and parse SPS state from the supplied buffer. // Unpack RBSP and parse SPS state from the supplied buffer.
rtc::Optional<SpsParser::SpsState> SpsParser::ParseSps(const uint8_t* data, absl::optional<SpsParser::SpsState> SpsParser::ParseSps(const uint8_t* data,
size_t length) { size_t length) {
std::vector<uint8_t> unpacked_buffer = H264::ParseRbsp(data, length); std::vector<uint8_t> unpacked_buffer = H264::ParseRbsp(data, length);
rtc::BitBuffer bit_buffer(unpacked_buffer.data(), unpacked_buffer.size()); rtc::BitBuffer bit_buffer(unpacked_buffer.data(), unpacked_buffer.size());
return ParseSpsUpToVui(&bit_buffer); return ParseSpsUpToVui(&bit_buffer);
} }
rtc::Optional<SpsParser::SpsState> SpsParser::ParseSpsUpToVui( absl::optional<SpsParser::SpsState> SpsParser::ParseSpsUpToVui(
rtc::BitBuffer* buffer) { rtc::BitBuffer* buffer) {
// Now, we need to use a bit buffer to parse through the actual AVC SPS // Now, we need to use a bit buffer to parse through the actual AVC SPS
// format. See Section 7.3.2.1.1 ("Sequence parameter set data syntax") of the // format. See Section 7.3.2.1.1 ("Sequence parameter set data syntax") of the

6
common_video/h264/sps_parser.h
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@ -11,7 +11,7 @@
#ifndef COMMON_VIDEO_H264_SPS_PARSER_H_ #ifndef COMMON_VIDEO_H264_SPS_PARSER_H_
#define COMMON_VIDEO_H264_SPS_PARSER_H_ #define COMMON_VIDEO_H264_SPS_PARSER_H_
#include "api/optional.h" #include "absl/types/optional.h"
namespace rtc { namespace rtc {
class BitBuffer; class BitBuffer;
@ -41,12 +41,12 @@ class SpsParser {
}; };
// Unpack RBSP and parse SPS state from the supplied buffer. // Unpack RBSP and parse SPS state from the supplied buffer.
static rtc::Optional<SpsState> ParseSps(const uint8_t* data, size_t length); static absl::optional<SpsState> ParseSps(const uint8_t* data, size_t length);
protected: protected:
// Parse the SPS state, up till the VUI part, for a bit buffer where RBSP // Parse the SPS state, up till the VUI part, for a bit buffer where RBSP
// decoding has already been performed. // decoding has already been performed.
static rtc::Optional<SpsState> ParseSpsUpToVui(rtc::BitBuffer* buffer); static absl::optional<SpsState> ParseSpsUpToVui(rtc::BitBuffer* buffer);
}; };
} // namespace webrtc } // namespace webrtc

2
common_video/h264/sps_parser_unittest.cc
View File

@ -112,7 +112,7 @@ class H264SpsParserTest : public ::testing::Test {
H264SpsParserTest() {} H264SpsParserTest() {}
virtual ~H264SpsParserTest() {} virtual ~H264SpsParserTest() {}
rtc::Optional<SpsParser::SpsState> sps_; absl::optional<SpsParser::SpsState> sps_;
}; };
TEST_F(H264SpsParserTest, TestSampleSPSHdLandscape) { TEST_F(H264SpsParserTest, TestSampleSPSHdLandscape) {

4
common_video/h264/sps_vui_rewriter.cc
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@ -72,13 +72,13 @@ bool CopyRemainingBits(rtc::BitBuffer* source,
SpsVuiRewriter::ParseResult SpsVuiRewriter::ParseAndRewriteSps( SpsVuiRewriter::ParseResult SpsVuiRewriter::ParseAndRewriteSps(
const uint8_t* buffer, const uint8_t* buffer,
size_t length, size_t length,
rtc::Optional<SpsParser::SpsState>* sps, absl::optional<SpsParser::SpsState>* sps,
rtc::Buffer* destination) { rtc::Buffer* destination) {
// Create temporary RBSP decoded buffer of the payload (exlcuding the // Create temporary RBSP decoded buffer of the payload (exlcuding the
// leading nalu type header byte (the SpsParser uses only the payload). // leading nalu type header byte (the SpsParser uses only the payload).
std::vector<uint8_t> rbsp_buffer = H264::ParseRbsp(buffer, length); std::vector<uint8_t> rbsp_buffer = H264::ParseRbsp(buffer, length);
rtc::BitBuffer source_buffer(rbsp_buffer.data(), rbsp_buffer.size()); rtc::BitBuffer source_buffer(rbsp_buffer.data(), rbsp_buffer.size());
rtc::Optional<SpsParser::SpsState> sps_state = absl::optional<SpsParser::SpsState> sps_state =
SpsParser::ParseSpsUpToVui(&source_buffer); SpsParser::ParseSpsUpToVui(&source_buffer);
if (!sps_state) if (!sps_state)
return ParseResult::kFailure; return ParseResult::kFailure;

7
common_video/h264/sps_vui_rewriter.h
View File

@ -12,7 +12,7 @@
#ifndef COMMON_VIDEO_H264_SPS_VUI_REWRITER_H_ #ifndef COMMON_VIDEO_H264_SPS_VUI_REWRITER_H_
#define COMMON_VIDEO_H264_SPS_VUI_REWRITER_H_ #define COMMON_VIDEO_H264_SPS_VUI_REWRITER_H_
#include "api/optional.h" #include "absl/types/optional.h"
#include "common_video/h264/sps_parser.h" #include "common_video/h264/sps_parser.h"
#include "rtc_base/buffer.h" #include "rtc_base/buffer.h"
@ -43,9 +43,10 @@ class SpsVuiRewriter : private SpsParser {
// SPS state. This function assumes that any previous headers // SPS state. This function assumes that any previous headers
// (NALU start, type, Stap-A, etc) have already been parsed and that RBSP // (NALU start, type, Stap-A, etc) have already been parsed and that RBSP
// decoding has been performed. // decoding has been performed.
static ParseResult ParseAndRewriteSps(const uint8_t* buffer, static ParseResult ParseAndRewriteSps(
const uint8_t* buffer,
size_t length, size_t length,
rtc::Optional<SpsParser::SpsState>* sps, absl::optional<SpsParser::SpsState>* sps,
rtc::Buffer* destination); rtc::Buffer* destination);
}; };

2
common_video/h264/sps_vui_rewriter_unittest.cc
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@ -159,7 +159,7 @@ void TestSps(SpsMode mode, SpsVuiRewriter::ParseResult expected_parse_result) {
index.payload_start_offset += H264::kNaluTypeSize; index.payload_start_offset += H264::kNaluTypeSize;
index.payload_size -= H264::kNaluTypeSize; index.payload_size -= H264::kNaluTypeSize;
rtc::Optional<SpsParser::SpsState> sps; absl::optional<SpsParser::SpsState> sps;
rtc::Buffer out_buffer; rtc::Buffer out_buffer;
SpsVuiRewriter::ParseResult result = SpsVuiRewriter::ParseResult result =
SpsVuiRewriter::ParseAndRewriteSps(&buffer[index.payload_start_offset], SpsVuiRewriter::ParseAndRewriteSps(&buffer[index.payload_start_offset],

2
common_video/include/bitrate_adjuster.h
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@ -44,7 +44,7 @@ class BitrateAdjuster {
uint32_t GetAdjustedBitrateBps() const; uint32_t GetAdjustedBitrateBps() const;
// Returns what we think the current bitrate is. // Returns what we think the current bitrate is.
rtc::Optional<uint32_t> GetEstimatedBitrateBps(); absl::optional<uint32_t> GetEstimatedBitrateBps();
// This should be called after each frame is encoded. The timestamp at which // This should be called after each frame is encoded. The timestamp at which
// it is called is used to estimate the output bitrate of the encoder. // it is called is used to estimate the output bitrate of the encoder.

2
common_video/incoming_video_stream.cc
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@ -69,7 +69,7 @@ void IncomingVideoStream::OnFrame(const VideoFrame& video_frame) {
void IncomingVideoStream::Dequeue() { void IncomingVideoStream::Dequeue() {
TRACE_EVENT0("webrtc", "IncomingVideoStream::Dequeue"); TRACE_EVENT0("webrtc", "IncomingVideoStream::Dequeue");
RTC_DCHECK(incoming_render_queue_.IsCurrent()); RTC_DCHECK(incoming_render_queue_.IsCurrent());
rtc::Optional<VideoFrame> frame_to_render = render_buffers_.FrameToRender(); absl::optional<VideoFrame> frame_to_render = render_buffers_.FrameToRender();
if (frame_to_render) if (frame_to_render)
callback_->OnFrame(*frame_to_render); callback_->OnFrame(*frame_to_render);

4
common_video/video_render_frames.cc
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@ -73,8 +73,8 @@ int32_t VideoRenderFrames::AddFrame(VideoFrame&& new_frame) {
return static_cast<int32_t>(incoming_frames_.size()); return static_cast<int32_t>(incoming_frames_.size());
} }
rtc::Optional<VideoFrame> VideoRenderFrames::FrameToRender() { absl::optional<VideoFrame> VideoRenderFrames::FrameToRender() {
rtc::Optional<VideoFrame> render_frame; absl::optional<VideoFrame> render_frame;
// Get the newest frame that can be released for rendering. // Get the newest frame that can be released for rendering.
while (!incoming_frames_.empty() && TimeToNextFrameRelease() <= 0) { while (!incoming_frames_.empty() && TimeToNextFrameRelease() <= 0) {
render_frame = std::move(incoming_frames_.front()); render_frame = std::move(incoming_frames_.front());

4
common_video/video_render_frames.h
View File

@ -15,7 +15,7 @@
#include <list> #include <list>
#include "api/optional.h" #include "absl/types/optional.h"
#include "api/video/video_frame.h" #include "api/video/video_frame.h"
namespace webrtc { namespace webrtc {
@ -30,7 +30,7 @@ class VideoRenderFrames {
int32_t AddFrame(VideoFrame&& new_frame); int32_t AddFrame(VideoFrame&& new_frame);
// Get a frame for rendering, or false if it's not time to render. // Get a frame for rendering, or false if it's not time to render.
rtc::Optional<VideoFrame> FrameToRender(); absl::optional<VideoFrame> FrameToRender();
// Returns the number of ms to next frame to render // Returns the number of ms to next frame to render
uint32_t TimeToNextFrameRelease(); uint32_t TimeToNextFrameRelease();

2
examples/peerconnection/client/conductor.cc
View File

@ -270,7 +270,7 @@ void Conductor::OnMessageFromPeer(int peer_id, const std::string& message) {
} }
return; return;
} }
rtc::Optional<webrtc::SdpType> type_maybe = absl::optional<webrtc::SdpType> type_maybe =
webrtc::SdpTypeFromString(type_str); webrtc::SdpTypeFromString(type_str);
if (!type_maybe) { if (!type_maybe) {
RTC_LOG(LS_ERROR) << "Unknown SDP type: " << type_str; RTC_LOG(LS_ERROR) << "Unknown SDP type: " << type_str;

4
logging/rtc_event_log/rtc_event_log_parser.h
View File

@ -57,8 +57,8 @@ class ParsedRtcEventLog {
struct BweProbeResultEvent { struct BweProbeResultEvent {
uint64_t timestamp; uint64_t timestamp;
uint32_t id; uint32_t id;
rtc::Optional<uint64_t> bitrate_bps; absl::optional<uint64_t> bitrate_bps;
rtc::Optional<ProbeFailureReason> failure_reason; absl::optional<ProbeFailureReason> failure_reason;
}; };
struct BweDelayBasedUpdate { struct BweDelayBasedUpdate {

4
logging/rtc_event_log/rtc_event_log_unittest.cc
View File

@ -748,8 +748,8 @@ TEST(RtcEventLogTest, CircularBufferKeepsMostRecentEvents) {
EXPECT_GT(parsed_log.GetNumberOfEvents(), 2u); EXPECT_GT(parsed_log.GetNumberOfEvents(), 2u);
RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0); RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0);
rtc::Optional<int64_t> last_timestamp; absl::optional<int64_t> last_timestamp;
rtc::Optional<uint32_t> last_ssrc; absl::optional<uint32_t> last_ssrc;
for (size_t i = 1; i < parsed_log.GetNumberOfEvents() - 1; i++) { for (size_t i = 1; i < parsed_log.GetNumberOfEvents() - 1; i++) {
EXPECT_EQ(parsed_log.GetEventType(i), EXPECT_EQ(parsed_log.GetEventType(i),
ParsedRtcEventLogNew::EventType::AUDIO_PLAYOUT_EVENT); ParsedRtcEventLogNew::EventType::AUDIO_PLAYOUT_EVENT);

4
modules/BUILD.gn
View File

@ -37,7 +37,7 @@ rtc_source_set("module_api_public") {
deps = [ deps = [
"..:webrtc_common", "..:webrtc_common",
"../:typedefs", "../:typedefs",
"../api:optional", "//third_party/abseil-cpp/absl/types:optional",
] ]
} }
@ -53,13 +53,13 @@ rtc_source_set("module_api") {
"..:webrtc_common", "..:webrtc_common",
"../:typedefs", "../:typedefs",
"../api:libjingle_peerconnection_api", "../api:libjingle_peerconnection_api",
"../api:optional",
"../api/transport:network_control", "../api/transport:network_control",
"../api/video:video_frame", "../api/video:video_frame",
"../api/video:video_frame_i420", "../api/video:video_frame_i420",
"../rtc_base:deprecation", "../rtc_base:deprecation",
"../rtc_base:rtc_base_approved", "../rtc_base:rtc_base_approved",
"video_coding:codec_globals_headers", "video_coding:codec_globals_headers",
"//third_party/abseil-cpp/absl/types:optional",
] ]
} }

4
modules/audio_device/BUILD.gn
View File

@ -204,9 +204,9 @@ rtc_source_set("audio_device_module_from_input_and_output") {
":audio_device_api", ":audio_device_api",
":audio_device_buffer", ":audio_device_buffer",
":windows_core_audio_utility", ":windows_core_audio_utility",
"../../api:optional",
"../../rtc_base:checks", "../../rtc_base:checks",
"../../rtc_base:rtc_base_approved", "../../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/types:optional",
] ]
} }
} }
@ -483,7 +483,6 @@ if (rtc_include_tests) {
":audio_device_impl", ":audio_device_impl",
":mock_audio_device", ":mock_audio_device",
"../../api:array_view", "../../api:array_view",
"../../api:optional",
"../../common_audio", "../../common_audio",
"../../rtc_base:checks", "../../rtc_base:checks",
"../../rtc_base:rtc_base_approved", "../../rtc_base:rtc_base_approved",
@ -491,6 +490,7 @@ if (rtc_include_tests) {
"../../test:fileutils", "../../test:fileutils",
"../../test:test_support", "../../test:test_support",
"../utility:utility", "../utility:utility",
"//third_party/abseil-cpp/absl/types:optional",
] ]
if (is_linux || is_mac || is_win) { if (is_linux || is_mac || is_win) {
sources += [ "audio_device_unittest.cc" ] sources += [ "audio_device_unittest.cc" ]

4
modules/audio_device/audio_device_unittest.cc
View File

@ -13,8 +13,8 @@
#include <memory> #include <memory>
#include <numeric> #include <numeric>
#include "absl/types/optional.h"
#include "api/array_view.h" #include "api/array_view.h"
#include "api/optional.h"
#include "modules/audio_device/audio_device_impl.h" #include "modules/audio_device/audio_device_impl.h"
#include "modules/audio_device/include/audio_device.h" #include "modules/audio_device/include/audio_device.h"
#include "modules/audio_device/include/mock_audio_transport.h" #include "modules/audio_device/include/mock_audio_transport.h"
@ -294,7 +294,7 @@ class LatencyAudioStream : public AudioStream {
rtc::ThreadChecker read_thread_checker_; rtc::ThreadChecker read_thread_checker_;
rtc::ThreadChecker write_thread_checker_; rtc::ThreadChecker write_thread_checker_;
rtc::Optional<int64_t> pulse_time_ RTC_GUARDED_BY(lock_); absl::optional<int64_t> pulse_time_ RTC_GUARDED_BY(lock_);
std::vector<int> latencies_ RTC_GUARDED_BY(race_checker_); std::vector<int> latencies_ RTC_GUARDED_BY(race_checker_);
size_t read_count_ RTC_GUARDED_BY(read_thread_checker_) = 0; size_t read_count_ RTC_GUARDED_BY(read_thread_checker_) = 0;
size_t write_count_ RTC_GUARDED_BY(write_thread_checker_) = 0; size_t write_count_ RTC_GUARDED_BY(write_thread_checker_) = 0;

6
modules/audio_device/win/core_audio_input_win.cc
View File

@ -315,10 +315,10 @@ bool CoreAudioInput::OnDataCallback(uint64_t device_frequency) {
return true; return true;
} }
rtc::Optional<int> CoreAudioInput::EstimateLatencyMillis( absl::optional<int> CoreAudioInput::EstimateLatencyMillis(
uint64_t capture_time_100ns) { uint64_t capture_time_100ns) {
if (!qpc_to_100ns_) { if (!qpc_to_100ns_) {
return rtc::nullopt; return absl::nullopt;
} }
// Input parameter |capture_time_100ns| contains the performance counter at // Input parameter |capture_time_100ns| contains the performance counter at
// the time that the audio endpoint device recorded the device position of // the time that the audio endpoint device recorded the device position of
@ -329,7 +329,7 @@ rtc::Optional<int> CoreAudioInput::EstimateLatencyMillis(
// - subtracting |capture_time_100ns| from now_time_100ns. // - subtracting |capture_time_100ns| from now_time_100ns.
LARGE_INTEGER perf_counter_now = {}; LARGE_INTEGER perf_counter_now = {};
if (!::QueryPerformanceCounter(&perf_counter_now)) { if (!::QueryPerformanceCounter(&perf_counter_now)) {
return rtc::nullopt; return absl::nullopt;
} }
uint64_t qpc_now_raw = perf_counter_now.QuadPart; uint64_t qpc_now_raw = perf_counter_now.QuadPart;
uint64_t now_time_100ns = qpc_now_raw * (*qpc_to_100ns_); uint64_t now_time_100ns = qpc_now_raw * (*qpc_to_100ns_);

6
modules/audio_device/win/core_audio_input_win.h
View File

@ -14,7 +14,7 @@
#include <memory> #include <memory>
#include <string> #include <string>
#include "api/optional.h" #include "absl/types/optional.h"
#include "modules/audio_device/win/audio_device_module_win.h" #include "modules/audio_device/win/audio_device_module_win.h"
#include "modules/audio_device/win/core_audio_base_win.h" #include "modules/audio_device/win/core_audio_base_win.h"
@ -53,11 +53,11 @@ class CoreAudioInput final : public CoreAudioBase, public AudioInput {
private: private:
bool OnDataCallback(uint64_t device_frequency); bool OnDataCallback(uint64_t device_frequency);
rtc::Optional<int> EstimateLatencyMillis(uint64_t capture_time_100ns); absl::optional<int> EstimateLatencyMillis(uint64_t capture_time_100ns);
std::unique_ptr<FineAudioBuffer> fine_audio_buffer_; std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
Microsoft::WRL::ComPtr<IAudioCaptureClient> audio_capture_client_; Microsoft::WRL::ComPtr<IAudioCaptureClient> audio_capture_client_;
rtc::Optional<double> qpc_to_100ns_; absl::optional<double> qpc_to_100ns_;
}; };
} // namespace webrtc_win } // namespace webrtc_win

2
modules/include/module_common_types.h
View File

@ -17,7 +17,7 @@
#include <algorithm> #include <algorithm>
#include <limits> #include <limits>
#include "api/optional.h" #include "absl/types/optional.h"
#include "api/rtp_headers.h" #include "api/rtp_headers.h"
#include "api/transport/network_types.h" #include "api/transport/network_types.h"
#include "api/video/video_rotation.h" #include "api/video/video_rotation.h"

4
modules/include/module_common_types_public.h
View File

@ -13,7 +13,7 @@
#include <limits> #include <limits>
#include "api/optional.h" #include "absl/types/optional.h"
#include "typedefs.h" // NOLINT(build/include) #include "typedefs.h" // NOLINT(build/include)
namespace webrtc { namespace webrtc {
@ -78,7 +78,7 @@ class Unwrapper {
} }
private: private:
rtc::Optional<int64_t> last_value_; absl::optional<int64_t> last_value_;
}; };
using SequenceNumberUnwrapper = Unwrapper<uint16_t>; using SequenceNumberUnwrapper = Unwrapper<uint16_t>;

2
sdk/BUILD.gn
View File

@ -304,12 +304,12 @@ if (is_ios || is_mac) {
":videoframebuffer_objc", ":videoframebuffer_objc",
":videosource_objc", ":videosource_objc",
"../api:libjingle_peerconnection_api", "../api:libjingle_peerconnection_api",
"../api:optional",
"../api/video:video_frame", "../api/video:video_frame",
"../common_video", "../common_video",
"../media:rtc_media_base", "../media:rtc_media_base",
"../rtc_base:checks", "../rtc_base:checks",
"../rtc_base:rtc_base", "../rtc_base:rtc_base",
"//third_party/abseil-cpp/absl/types:optional",
] ]
configs += [ configs += [

12
sdk/android/BUILD.gn
View File

@ -87,11 +87,11 @@ rtc_source_set("base_jni") {
":internal_jni", ":internal_jni",
":native_api_jni", ":native_api_jni",
"../../api:libjingle_peerconnection_api", "../../api:libjingle_peerconnection_api",
"../../api:optional",
"../../rtc_base:checks", "../../rtc_base:checks",
"../../rtc_base:rtc_base", "../../rtc_base:rtc_base",
"../../rtc_base:rtc_base_approved", "../../rtc_base:rtc_base_approved",
"../../system_wrappers:metrics_api", "../../system_wrappers:metrics_api",
"//third_party/abseil-cpp/absl/types:optional",
] ]
} }
@ -145,11 +145,11 @@ rtc_source_set("audio_device_module_base") {
":base_jni", ":base_jni",
":generated_audio_device_module_base_jni", ":generated_audio_device_module_base_jni",
":native_api_jni", ":native_api_jni",
"../../api:optional",
"../../modules/audio_device:audio_device_buffer", "../../modules/audio_device:audio_device_buffer",
"../../rtc_base:checks", "../../rtc_base:checks",
"../../rtc_base:rtc_base_approved", "../../rtc_base:rtc_base_approved",
"../../system_wrappers:metrics_api", "../../system_wrappers:metrics_api",
"//third_party/abseil-cpp/absl/types:optional",
] ]
} }
@ -170,13 +170,13 @@ if (rtc_enable_android_aaudio) {
":audio_device_module_base", ":audio_device_module_base",
":base_jni", ":base_jni",
"../../api:array_view", "../../api:array_view",
"../../api:optional",
"../../modules/audio_device:audio_device", "../../modules/audio_device:audio_device",
"../../modules/audio_device:audio_device_buffer", "../../modules/audio_device:audio_device_buffer",
"../../rtc_base:checks", "../../rtc_base:checks",
"../../rtc_base:rtc_base", "../../rtc_base:rtc_base",
"../../rtc_base:rtc_base_approved", "../../rtc_base:rtc_base_approved",
"../../system_wrappers", "../../system_wrappers",
"//third_party/abseil-cpp/absl/types:optional",
] ]
} }
} }
@ -196,11 +196,11 @@ rtc_source_set("opensles_audio_device_module") {
":audio_device_module_base", ":audio_device_module_base",
":base_jni", ":base_jni",
"../../api:array_view", "../../api:array_view",
"../../api:optional",
"../../modules/audio_device:audio_device", "../../modules/audio_device:audio_device",
"../../modules/audio_device:audio_device_buffer", "../../modules/audio_device:audio_device_buffer",
"../../rtc_base:checks", "../../rtc_base:checks",
"../../rtc_base:rtc_base_approved", "../../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/types:optional",
] ]
} }
@ -217,12 +217,12 @@ rtc_source_set("java_audio_device_module") {
":audio_device_module_base", ":audio_device_module_base",
":base_jni", ":base_jni",
":generated_java_audio_device_module_native_jni", ":generated_java_audio_device_module_native_jni",
"../../api:optional",
"../../modules/audio_device:audio_device", "../../modules/audio_device:audio_device",
"../../modules/audio_device:audio_device_buffer", "../../modules/audio_device:audio_device_buffer",
"../../rtc_base:checks", "../../rtc_base:checks",
"../../rtc_base:rtc_base_approved", "../../rtc_base:rtc_base_approved",
"../../system_wrappers:metrics_api", "../../system_wrappers:metrics_api",
"//third_party/abseil-cpp/absl/types:optional",
] ]
} }
@ -1148,9 +1148,9 @@ rtc_static_library("native_api_jni") {
":generated_external_classes_jni", ":generated_external_classes_jni",
":generated_native_api_jni", ":generated_native_api_jni",
":internal_jni", ":internal_jni",
"//api:optional",
"//rtc_base:checks", "//rtc_base:checks",
"//rtc_base:rtc_base_approved", "//rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/types:optional",
] ]
} }

12
sdk/android/native_api/jni/java_types.cc
View File

@ -125,18 +125,18 @@ int64_t JavaToNativeLong(JNIEnv* env, const JavaRef<jobject>& j_long) {
return JNI_Long::Java_Long_longValue(env, j_long); return JNI_Long::Java_Long_longValue(env, j_long);
} }
rtc::Optional<bool> JavaToNativeOptionalBool(JNIEnv* jni, absl::optional<bool> JavaToNativeOptionalBool(JNIEnv* jni,
const JavaRef<jobject>& boolean) { const JavaRef<jobject>& boolean) {
if (IsNull(jni, boolean)) if (IsNull(jni, boolean))
return rtc::nullopt; return absl::nullopt;
return JNI_Boolean::Java_Boolean_booleanValue(jni, boolean); return JNI_Boolean::Java_Boolean_booleanValue(jni, boolean);
} }
rtc::Optional<int32_t> JavaToNativeOptionalInt( absl::optional<int32_t> JavaToNativeOptionalInt(
JNIEnv* jni, JNIEnv* jni,
const JavaRef<jobject>& integer) { const JavaRef<jobject>& integer) {
if (IsNull(jni, integer)) if (IsNull(jni, integer))
return rtc::nullopt; return absl::nullopt;
return JNI_Integer::Java_Integer_intValue(jni, integer); return JNI_Integer::Java_Integer_intValue(jni, integer);
} }
@ -196,13 +196,13 @@ ScopedJavaLocalRef<jstring> NativeToJavaString(JNIEnv* jni,
ScopedJavaLocalRef<jobject> NativeToJavaInteger( ScopedJavaLocalRef<jobject> NativeToJavaInteger(
JNIEnv* jni, JNIEnv* jni,
const rtc::Optional<int32_t>& optional_int) { const absl::optional<int32_t>& optional_int) {
return optional_int ? NativeToJavaInteger(jni, *optional_int) : nullptr; return optional_int ? NativeToJavaInteger(jni, *optional_int) : nullptr;
} }
ScopedJavaLocalRef<jstring> NativeToJavaString( ScopedJavaLocalRef<jstring> NativeToJavaString(
JNIEnv* jni, JNIEnv* jni,
const rtc::Optional<std::string>& str) { const absl::optional<std::string>& str) {
return str ? NativeToJavaString(jni, *str) : nullptr; return str ? NativeToJavaString(jni, *str) : nullptr;
} }

11
sdk/android/native_api/jni/java_types.h
View File

@ -22,7 +22,7 @@
#include <string> #include <string>
#include <vector> #include <vector>
#include "api/optional.h" #include "absl/types/optional.h"
#include "rtc_base/checks.h" #include "rtc_base/checks.h"
#include "rtc_base/thread_checker.h" #include "rtc_base/thread_checker.h"
#include "sdk/android/native_api/jni/scoped_java_ref.h" #include "sdk/android/native_api/jni/scoped_java_ref.h"
@ -126,9 +126,10 @@ ScopedJavaLocalRef<jobject> GetJavaMapEntryValue(
int64_t JavaToNativeLong(JNIEnv* env, const JavaRef<jobject>& j_long); int64_t JavaToNativeLong(JNIEnv* env, const JavaRef<jobject>& j_long);
rtc::Optional<bool> JavaToNativeOptionalBool(JNIEnv* jni, absl::optional<bool> JavaToNativeOptionalBool(JNIEnv* jni,
const JavaRef<jobject>& boolean); const JavaRef<jobject>& boolean);
rtc::Optional<int32_t> JavaToNativeOptionalInt(JNIEnv* jni, absl::optional<int32_t> JavaToNativeOptionalInt(
JNIEnv* jni,
const JavaRef<jobject>& integer); const JavaRef<jobject>& integer);
// Given a (UTF-16) jstring return a new UTF-8 native string. // Given a (UTF-16) jstring return a new UTF-8 native string.
@ -196,10 +197,10 @@ ScopedJavaLocalRef<jstring> NativeToJavaString(JNIEnv* jni,
ScopedJavaLocalRef<jobject> NativeToJavaInteger( ScopedJavaLocalRef<jobject> NativeToJavaInteger(
JNIEnv* jni, JNIEnv* jni,
const rtc::Optional<int32_t>& optional_int); const absl::optional<int32_t>& optional_int);
ScopedJavaLocalRef<jstring> NativeToJavaString( ScopedJavaLocalRef<jstring> NativeToJavaString(
JNIEnv* jni, JNIEnv* jni,
const rtc::Optional<std::string>& str); const absl::optional<std::string>& str);
// Helper function for converting std::vector<T> into a Java array. // Helper function for converting std::vector<T> into a Java array.
template <typename T, typename Convert> template <typename T, typename Convert>

2
sdk/android/native_api/video/videosource.cc
View File

@ -74,7 +74,7 @@ class JavaVideoTrackSourceImpl : public JavaVideoTrackSourceInterface {
return android_video_track_source_->is_screencast(); return android_video_track_source_->is_screencast();
} }
rtc::Optional<bool> needs_denoising() const override { absl::optional<bool> needs_denoising() const override {
return android_video_track_source_->needs_denoising(); return android_video_track_source_->needs_denoising();
} }

6
sdk/android/src/jni/androidmediadecoder.cc
View File

@ -124,7 +124,7 @@ class MediaCodecVideoDecoder : public VideoDecoder, public rtc::MessageHandler {
int current_delay_time_ms_; // Overall delay time in the current second. int current_delay_time_ms_; // Overall delay time in the current second.
int32_t max_pending_frames_; // Maximum number of pending input frames. int32_t max_pending_frames_; // Maximum number of pending input frames.
H264BitstreamParser h264_bitstream_parser_; H264BitstreamParser h264_bitstream_parser_;
std::deque<rtc::Optional<uint8_t>> pending_frame_qps_; std::deque<absl::optional<uint8_t>> pending_frame_qps_;
// State that is constant for the lifetime of this object once the ctor // State that is constant for the lifetime of this object once the ctor
// returns. // returns.
@ -506,7 +506,7 @@ int32_t MediaCodecVideoDecoder::DecodeOnCodecThread(
// Save input image timestamps for later output. // Save input image timestamps for later output.
frames_received_++; frames_received_++;
current_bytes_ += inputImage._length; current_bytes_ += inputImage._length;
rtc::Optional<uint8_t> qp; absl::optional<uint8_t> qp;
if (codecType_ == kVideoCodecVP8) { if (codecType_ == kVideoCodecVP8) {
int qp_int; int qp_int;
if (vp8::GetQp(inputImage._buffer, inputImage._length, &qp_int)) { if (vp8::GetQp(inputImage._buffer, inputImage._length, &qp_int)) {
@ -743,7 +743,7 @@ bool MediaCodecVideoDecoder::DeliverPendingOutputs(JNIEnv* jni,
decoded_frame.set_timestamp(output_timestamps_ms); decoded_frame.set_timestamp(output_timestamps_ms);
decoded_frame.set_ntp_time_ms(output_ntp_timestamps_ms); decoded_frame.set_ntp_time_ms(output_ntp_timestamps_ms);
rtc::Optional<uint8_t> qp = pending_frame_qps_.front(); absl::optional<uint8_t> qp = pending_frame_qps_.front();
pending_frame_qps_.pop_front(); pending_frame_qps_.pop_front();
callback_->Decoded(decoded_frame, decode_time_ms, qp); callback_->Decoded(decoded_frame, decode_time_ms, qp);
} }

2
sdk/android/src/jni/androidmediaencoder.cc
View File

@ -347,7 +347,7 @@ int32_t MediaCodecVideoEncoder::InitEncode(const VideoCodec* codec_settings,
// Check allowed H.264 profile // Check allowed H.264 profile
profile_ = H264::Profile::kProfileBaseline; profile_ = H264::Profile::kProfileBaseline;
if (codec_type == kVideoCodecH264) { if (codec_type == kVideoCodecH264) {
const rtc::Optional<H264::ProfileLevelId> profile_level_id = const absl::optional<H264::ProfileLevelId> profile_level_id =
H264::ParseSdpProfileLevelId(codec_.params); H264::ParseSdpProfileLevelId(codec_.params);
RTC_DCHECK(profile_level_id); RTC_DCHECK(profile_level_id);
profile_ = profile_level_id->profile; profile_ = profile_level_id->profile;

2
sdk/android/src/jni/androidvideotracksource.cc
View File

@ -38,7 +38,7 @@ bool AndroidVideoTrackSource::is_screencast() const {
return is_screencast_; return is_screencast_;
} }
rtc::Optional<bool> AndroidVideoTrackSource::needs_denoising() const { absl::optional<bool> AndroidVideoTrackSource::needs_denoising() const {
return false; return false;
} }

2
sdk/android/src/jni/androidvideotracksource.h
View File

@ -37,7 +37,7 @@ class AndroidVideoTrackSource : public rtc::AdaptedVideoTrackSource {
// Indicates that the encoder should denoise video before encoding it. // Indicates that the encoder should denoise video before encoding it.
// If it is not set, the default configuration is used which is different // If it is not set, the default configuration is used which is different
// depending on video codec. // depending on video codec.
rtc::Optional<bool> needs_denoising() const override; absl::optional<bool> needs_denoising() const override;
// Called by the native capture observer // Called by the native capture observer
void SetState(SourceState state); void SetState(SourceState state);

12
sdk/android/src/jni/audio_device/aaudio_player.cc
View File

@ -135,16 +135,16 @@ int AAudioPlayer::SetSpeakerVolume(uint32_t volume) {
return -1; return -1;
} }
rtc::Optional<uint32_t> AAudioPlayer::SpeakerVolume() const { absl::optional<uint32_t> AAudioPlayer::SpeakerVolume() const {
return rtc::nullopt; return absl::nullopt;
} }
rtc::Optional<uint32_t> AAudioPlayer::MaxSpeakerVolume() const { absl::optional<uint32_t> AAudioPlayer::MaxSpeakerVolume() const {
return rtc::nullopt; return absl::nullopt;
} }
rtc::Optional<uint32_t> AAudioPlayer::MinSpeakerVolume() const { absl::optional<uint32_t> AAudioPlayer::MinSpeakerVolume() const {
return rtc::nullopt; return absl::nullopt;
} }
void AAudioPlayer::OnErrorCallback(aaudio_result_t error) { void AAudioPlayer::OnErrorCallback(aaudio_result_t error) {

8
sdk/android/src/jni/audio_device/aaudio_player.h
View File

@ -14,7 +14,7 @@
#include <aaudio/AAudio.h> #include <aaudio/AAudio.h>
#include <memory> #include <memory>
#include "api/optional.h" #include "absl/types/optional.h"
#include "modules/audio_device/audio_device_buffer.h" #include "modules/audio_device/audio_device_buffer.h"
#include "modules/audio_device/include/audio_device_defines.h" #include "modules/audio_device/include/audio_device_defines.h"
#include "rtc_base/messagehandler.h" #include "rtc_base/messagehandler.h"
@ -73,9 +73,9 @@ class AAudioPlayer final : public AudioOutput,
// Not implemented in AAudio. // Not implemented in AAudio.
bool SpeakerVolumeIsAvailable() override; bool SpeakerVolumeIsAvailable() override;
int SetSpeakerVolume(uint32_t volume) override; int SetSpeakerVolume(uint32_t volume) override;
rtc::Optional<uint32_t> SpeakerVolume() const override; absl::optional<uint32_t> SpeakerVolume() const override;
rtc::Optional<uint32_t> MaxSpeakerVolume() const override; absl::optional<uint32_t> MaxSpeakerVolume() const override;
rtc::Optional<uint32_t> MinSpeakerVolume() const override; absl::optional<uint32_t> MinSpeakerVolume() const override;
protected: protected:
// AAudioObserverInterface implementation. // AAudioObserverInterface implementation.

6
sdk/android/src/jni/audio_device/audio_device_module.cc
View File

@ -341,7 +341,7 @@ class AndroidAudioDeviceModule : public AudioDeviceModule {
RTC_LOG(INFO) << __FUNCTION__; RTC_LOG(INFO) << __FUNCTION__;
if (!initialized_) if (!initialized_)
return -1; return -1;
rtc::Optional<uint32_t> volume = output_->SpeakerVolume(); absl::optional<uint32_t> volume = output_->SpeakerVolume();
if (!volume) if (!volume)
return -1; return -1;
*output_volume = *volume; *output_volume = *volume;
@ -353,7 +353,7 @@ class AndroidAudioDeviceModule : public AudioDeviceModule {
RTC_LOG(INFO) << __FUNCTION__; RTC_LOG(INFO) << __FUNCTION__;
if (!initialized_) if (!initialized_)
return -1; return -1;
rtc::Optional<uint32_t> max_volume = output_->MaxSpeakerVolume(); absl::optional<uint32_t> max_volume = output_->MaxSpeakerVolume();
if (!max_volume) if (!max_volume)
return -1; return -1;
*output_max_volume = *max_volume; *output_max_volume = *max_volume;
@ -364,7 +364,7 @@ class AndroidAudioDeviceModule : public AudioDeviceModule {
RTC_LOG(INFO) << __FUNCTION__; RTC_LOG(INFO) << __FUNCTION__;
if (!initialized_) if (!initialized_)
return -1; return -1;
rtc::Optional<uint32_t> min_volume = output_->MinSpeakerVolume(); absl::optional<uint32_t> min_volume = output_->MinSpeakerVolume();
if (!min_volume) if (!min_volume)
return -1; return -1;
*output_min_volume = *min_volume; *output_min_volume = *min_volume;

8
sdk/android/src/jni/audio_device/audio_device_module.h
View File

@ -13,7 +13,7 @@
#include <memory> #include <memory>
#include "api/optional.h" #include "absl/types/optional.h"
#include "modules/audio_device/audio_device_buffer.h" #include "modules/audio_device/audio_device_buffer.h"
#include "sdk/android/native_api/jni/scoped_java_ref.h" #include "sdk/android/native_api/jni/scoped_java_ref.h"
@ -58,9 +58,9 @@ class AudioOutput {
virtual bool Playing() const = 0; virtual bool Playing() const = 0;
virtual bool SpeakerVolumeIsAvailable() = 0; virtual bool SpeakerVolumeIsAvailable() = 0;
virtual int SetSpeakerVolume(uint32_t volume) = 0; virtual int SetSpeakerVolume(uint32_t volume) = 0;
virtual rtc::Optional<uint32_t> SpeakerVolume() const = 0; virtual absl::optional<uint32_t> SpeakerVolume() const = 0;
virtual rtc::Optional<uint32_t> MaxSpeakerVolume() const = 0; virtual absl::optional<uint32_t> MaxSpeakerVolume() const = 0;
virtual rtc::Optional<uint32_t> MinSpeakerVolume() const = 0; virtual absl::optional<uint32_t> MinSpeakerVolume() const = 0;
virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0; virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
}; };

6
sdk/android/src/jni/audio_device/audio_track_jni.cc
View File

@ -144,17 +144,17 @@ int AudioTrackJni::SetSpeakerVolume(uint32_t volume) {
: -1; : -1;
} }
rtc::Optional<uint32_t> AudioTrackJni::MaxSpeakerVolume() const { absl::optional<uint32_t> AudioTrackJni::MaxSpeakerVolume() const {
RTC_DCHECK(thread_checker_.CalledOnValidThread()); RTC_DCHECK(thread_checker_.CalledOnValidThread());
return Java_WebRtcAudioTrack_getStreamMaxVolume(env_, j_audio_track_); return Java_WebRtcAudioTrack_getStreamMaxVolume(env_, j_audio_track_);
} }
rtc::Optional<uint32_t> AudioTrackJni::MinSpeakerVolume() const { absl::optional<uint32_t> AudioTrackJni::MinSpeakerVolume() const {
RTC_DCHECK(thread_checker_.CalledOnValidThread()); RTC_DCHECK(thread_checker_.CalledOnValidThread());
return 0; return 0;
} }
rtc::Optional<uint32_t> AudioTrackJni::SpeakerVolume() const { absl::optional<uint32_t> AudioTrackJni::SpeakerVolume() const {
RTC_DCHECK(thread_checker_.CalledOnValidThread()); RTC_DCHECK(thread_checker_.CalledOnValidThread());
const uint32_t volume = const uint32_t volume =
Java_WebRtcAudioTrack_getStreamVolume(env_, j_audio_track_); Java_WebRtcAudioTrack_getStreamVolume(env_, j_audio_track_);

8
sdk/android/src/jni/audio_device/audio_track_jni.h
View File

@ -14,7 +14,7 @@
#include <jni.h> #include <jni.h>
#include <memory> #include <memory>
#include "api/optional.h" #include "absl/types/optional.h"
#include "modules/audio_device/audio_device_buffer.h" #include "modules/audio_device/audio_device_buffer.h"
#include "modules/audio_device/include/audio_device_defines.h" #include "modules/audio_device/include/audio_device_defines.h"
#include "rtc_base/thread_checker.h" #include "rtc_base/thread_checker.h"
@ -62,9 +62,9 @@ class AudioTrackJni : public AudioOutput {
bool SpeakerVolumeIsAvailable() override; bool SpeakerVolumeIsAvailable() override;
int SetSpeakerVolume(uint32_t volume) override; int SetSpeakerVolume(uint32_t volume) override;
rtc::Optional<uint32_t> SpeakerVolume() const override; absl::optional<uint32_t> SpeakerVolume() const override;
rtc::Optional<uint32_t> MaxSpeakerVolume() const override; absl::optional<uint32_t> MaxSpeakerVolume() const override;
rtc::Optional<uint32_t> MinSpeakerVolume() const override; absl::optional<uint32_t> MinSpeakerVolume() const override;
void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override; void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;

12
sdk/android/src/jni/audio_device/opensles_player.cc
View File

@ -182,16 +182,16 @@ int OpenSLESPlayer::SetSpeakerVolume(uint32_t volume) {
return -1; return -1;
} }
rtc::Optional<uint32_t> OpenSLESPlayer::SpeakerVolume() const { absl::optional<uint32_t> OpenSLESPlayer::SpeakerVolume() const {
return rtc::nullopt; return absl::nullopt;
} }
rtc::Optional<uint32_t> OpenSLESPlayer::MaxSpeakerVolume() const { absl::optional<uint32_t> OpenSLESPlayer::MaxSpeakerVolume() const {
return rtc::nullopt; return absl::nullopt;
} }
rtc::Optional<uint32_t> OpenSLESPlayer::MinSpeakerVolume() const { absl::optional<uint32_t> OpenSLESPlayer::MinSpeakerVolume() const {
return rtc::nullopt; return absl::nullopt;
} }
void OpenSLESPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) { void OpenSLESPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {

8
sdk/android/src/jni/audio_device/opensles_player.h
View File

@ -16,7 +16,7 @@
#include <SLES/OpenSLES_AndroidConfiguration.h> #include <SLES/OpenSLES_AndroidConfiguration.h>
#include <memory> #include <memory>
#include "api/optional.h" #include "absl/types/optional.h"
#include "modules/audio_device/audio_device_buffer.h" #include "modules/audio_device/audio_device_buffer.h"
#include "modules/audio_device/fine_audio_buffer.h" #include "modules/audio_device/fine_audio_buffer.h"
#include "modules/audio_device/include/audio_device_defines.h" #include "modules/audio_device/include/audio_device_defines.h"
@ -75,9 +75,9 @@ class OpenSLESPlayer : public AudioOutput {
bool SpeakerVolumeIsAvailable() override; bool SpeakerVolumeIsAvailable() override;
int SetSpeakerVolume(uint32_t volume) override; int SetSpeakerVolume(uint32_t volume) override;
rtc::Optional<uint32_t> SpeakerVolume() const override; absl::optional<uint32_t> SpeakerVolume() const override;
rtc::Optional<uint32_t> MaxSpeakerVolume() const override; absl::optional<uint32_t> MaxSpeakerVolume() const override;
rtc::Optional<uint32_t> MinSpeakerVolume() const override; absl::optional<uint32_t> MinSpeakerVolume() const override;
void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override; void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;

6
sdk/android/src/jni/pc/icecandidate.cc
View File

@ -207,13 +207,13 @@ PeerConnectionInterface::TlsCertPolicy JavaToNativeTlsCertPolicy(
return PeerConnectionInterface::kTlsCertPolicySecure; return PeerConnectionInterface::kTlsCertPolicySecure;
} }
rtc::Optional<rtc::AdapterType> JavaToNativeNetworkPreference( absl::optional<rtc::AdapterType> JavaToNativeNetworkPreference(
JNIEnv* jni, JNIEnv* jni,
const JavaRef<jobject>& j_network_preference) { const JavaRef<jobject>& j_network_preference) {
std::string enum_name = GetJavaEnumName(jni, j_network_preference); std::string enum_name = GetJavaEnumName(jni, j_network_preference);
if (enum_name == "UNKNOWN") if (enum_name == "UNKNOWN")
return rtc::nullopt; return absl::nullopt;
if (enum_name == "ETHERNET") if (enum_name == "ETHERNET")
return rtc::ADAPTER_TYPE_ETHERNET; return rtc::ADAPTER_TYPE_ETHERNET;
@ -231,7 +231,7 @@ rtc::Optional<rtc::AdapterType> JavaToNativeNetworkPreference(
return rtc::ADAPTER_TYPE_LOOPBACK; return rtc::ADAPTER_TYPE_LOOPBACK;
RTC_CHECK(false) << "Unexpected NetworkPreference enum_name " << enum_name; RTC_CHECK(false) << "Unexpected NetworkPreference enum_name " << enum_name;
return rtc::nullopt; return absl::nullopt;
} }
} // namespace jni } // namespace jni

2
sdk/android/src/jni/pc/icecandidate.h
View File

@ -75,7 +75,7 @@ PeerConnectionInterface::TlsCertPolicy JavaToNativeTlsCertPolicy(
JNIEnv* jni, JNIEnv* jni,
const JavaRef<jobject>& j_ice_server_tls_cert_policy); const JavaRef<jobject>& j_ice_server_tls_cert_policy);
rtc::Optional<rtc::AdapterType> JavaToNativeNetworkPreference( absl::optional<rtc::AdapterType> JavaToNativeNetworkPreference(
JNIEnv* jni, JNIEnv* jni,
const JavaRef<jobject>& j_network_preference); const JavaRef<jobject>& j_network_preference);

2
sdk/android/src/jni/pc/peerconnectionfactory.cc
View File

@ -428,7 +428,7 @@ static jlong JNI_PeerConnectionFactory_CreatePeerConnection(
if (key_type != rtc::KT_DEFAULT) { if (key_type != rtc::KT_DEFAULT) {
rtc::scoped_refptr<rtc::RTCCertificate> certificate = rtc::scoped_refptr<rtc::RTCCertificate> certificate =
rtc::RTCCertificateGenerator::GenerateCertificate( rtc::RTCCertificateGenerator::GenerateCertificate(
rtc::KeyParams(key_type), rtc::nullopt); rtc::KeyParams(key_type), absl::nullopt);
if (!certificate) { if (!certificate) {
RTC_LOG(LS_ERROR) << "Failed to generate certificate. KeyType: " RTC_LOG(LS_ERROR) << "Failed to generate certificate. KeyType: "
<< key_type; << key_type;

4
sdk/android/src/jni/pc/rtptransceiver.cc
View File

@ -89,7 +89,7 @@ ScopedJavaLocalRef<jstring> JNI_RtpTransceiver_GetMid(
JNIEnv* jni, JNIEnv* jni,
const base::android::JavaParamRef<jclass>&, const base::android::JavaParamRef<jclass>&,
jlong j_rtp_transceiver_pointer) { jlong j_rtp_transceiver_pointer) {
rtc::Optional<std::string> mid = absl::optional<std::string> mid =
reinterpret_cast<RtpTransceiverInterface*>(j_rtp_transceiver_pointer) reinterpret_cast<RtpTransceiverInterface*>(j_rtp_transceiver_pointer)
->mid(); ->mid();
return NativeToJavaString(jni, mid); return NativeToJavaString(jni, mid);
@ -133,7 +133,7 @@ ScopedJavaLocalRef<jobject> JNI_RtpTransceiver_CurrentDirection(
JNIEnv* jni, JNIEnv* jni,
const base::android::JavaParamRef<jclass>&, const base::android::JavaParamRef<jclass>&,
jlong j_rtp_transceiver_pointer) { jlong j_rtp_transceiver_pointer) {
rtc::Optional<RtpTransceiverDirection> direction = absl::optional<RtpTransceiverDirection> direction =
reinterpret_cast<RtpTransceiverInterface*>(j_rtp_transceiver_pointer) reinterpret_cast<RtpTransceiverInterface*>(j_rtp_transceiver_pointer)
->current_direction(); ->current_direction();
return direction ? NativeToJavaRtpTransceiverDirection(jni, *direction) return direction ? NativeToJavaRtpTransceiverDirection(jni, *direction)

2
sdk/android/src/jni/pc/sessiondescription.cc
View File

@ -27,7 +27,7 @@ std::unique_ptr<SessionDescriptionInterface> JavaToNativeSessionDescription(
jni, Java_SessionDescription_getTypeInCanonicalForm(jni, j_sdp)); jni, Java_SessionDescription_getTypeInCanonicalForm(jni, j_sdp));
std::string std_description = std::string std_description =
JavaToStdString(jni, Java_SessionDescription_getDescription(jni, j_sdp)); JavaToStdString(jni, Java_SessionDescription_getDescription(jni, j_sdp));
rtc::Optional<SdpType> sdp_type_maybe = SdpTypeFromString(std_type); absl::optional<SdpType> sdp_type_maybe = SdpTypeFromString(std_type);
if (!sdp_type_maybe) { if (!sdp_type_maybe) {
RTC_LOG(LS_ERROR) << "Unexpected SDP type: " << std_type; RTC_LOG(LS_ERROR) << "Unexpected SDP type: " << std_type;
return nullptr; return nullptr;

16
sdk/android/src/jni/videodecoderwrapper.cc
View File

@ -30,9 +30,9 @@ namespace {
const int64_t kNumRtpTicksPerMillisec = 90000 / rtc::kNumMillisecsPerSec; const int64_t kNumRtpTicksPerMillisec = 90000 / rtc::kNumMillisecsPerSec;
template <typename Dst, typename Src> template <typename Dst, typename Src>
inline rtc::Optional<Dst> cast_optional(const rtc::Optional<Src>& value) { inline absl::optional<Dst> cast_optional(const absl::optional<Src>& value) {
return value ? rtc::Optional<Dst>(rtc::dchecked_cast<Dst, Src>(*value)) return value ? absl::optional<Dst>(rtc::dchecked_cast<Dst, Src>(*value))
: rtc::nullopt; : absl::nullopt;
} }
} // namespace } // namespace
@ -106,7 +106,7 @@ int32_t VideoDecoderWrapper::Decode(
frame_extra_info.timestamp_rtp = input_image._timeStamp; frame_extra_info.timestamp_rtp = input_image._timeStamp;
frame_extra_info.timestamp_ntp = input_image.ntp_time_ms_; frame_extra_info.timestamp_ntp = input_image.ntp_time_ms_;
frame_extra_info.qp = frame_extra_info.qp =
qp_parsing_enabled_ ? ParseQP(input_image) : rtc::nullopt; qp_parsing_enabled_ ? ParseQP(input_image) : absl::nullopt;
{ {
rtc::CritScope cs(&frame_extra_infos_lock_); rtc::CritScope cs(&frame_extra_infos_lock_);
frame_extra_infos_.push_back(frame_extra_info); frame_extra_infos_.push_back(frame_extra_info);
@ -183,10 +183,10 @@ void VideoDecoderWrapper::OnDecodedFrame(
JavaToNativeFrame(env, j_frame, frame_extra_info.timestamp_rtp); JavaToNativeFrame(env, j_frame, frame_extra_info.timestamp_rtp);
frame.set_ntp_time_ms(frame_extra_info.timestamp_ntp); frame.set_ntp_time_ms(frame_extra_info.timestamp_ntp);
rtc::Optional<int32_t> decoding_time_ms = absl::optional<int32_t> decoding_time_ms =
JavaToNativeOptionalInt(env, j_decode_time_ms); JavaToNativeOptionalInt(env, j_decode_time_ms);
rtc::Optional<uint8_t> decoder_qp = absl::optional<uint8_t> decoder_qp =
cast_optional<uint8_t, int32_t>(JavaToNativeOptionalInt(env, j_qp)); cast_optional<uint8_t, int32_t>(JavaToNativeOptionalInt(env, j_qp));
// If the decoder provides QP values itself, no need to parse the bitstream. // If the decoder provides QP values itself, no need to parse the bitstream.
// Enable QP parsing if decoder does not provide QP values itself. // Enable QP parsing if decoder does not provide QP values itself.
@ -226,13 +226,13 @@ int32_t VideoDecoderWrapper::HandleReturnCode(JNIEnv* jni,
return WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE; return WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
} }
rtc::Optional<uint8_t> VideoDecoderWrapper::ParseQP( absl::optional<uint8_t> VideoDecoderWrapper::ParseQP(
const EncodedImage& input_image) { const EncodedImage& input_image) {
if (input_image.qp_ != -1) { if (input_image.qp_ != -1) {
return input_image.qp_; return input_image.qp_;
} }
rtc::Optional<uint8_t> qp; absl::optional<uint8_t> qp;
switch (codec_settings_.codecType) { switch (codec_settings_.codecType) {
case kVideoCodecVP8: { case kVideoCodecVP8: {
int qp_int; int qp_int;

4
sdk/android/src/jni/videodecoderwrapper.h
View File

@ -66,7 +66,7 @@ class VideoDecoderWrapper : public VideoDecoder {
uint32_t timestamp_rtp; uint32_t timestamp_rtp;
int64_t timestamp_ntp; int64_t timestamp_ntp;
rtc::Optional<uint8_t> qp; absl::optional<uint8_t> qp;
FrameExtraInfo(); FrameExtraInfo();
FrameExtraInfo(const FrameExtraInfo&); FrameExtraInfo(const FrameExtraInfo&);
@ -82,7 +82,7 @@ class VideoDecoderWrapper : public VideoDecoder {
const char* method_name) const char* method_name)
RTC_RUN_ON(decoder_thread_checker_); RTC_RUN_ON(decoder_thread_checker_);
rtc::Optional<uint8_t> ParseQP(const EncodedImage& input_image) absl::optional<uint8_t> ParseQP(const EncodedImage& input_image)
RTC_RUN_ON(decoder_thread_checker_); RTC_RUN_ON(decoder_thread_checker_);
const ScopedJavaGlobalRef<jobject> decoder_; const ScopedJavaGlobalRef<jobject> decoder_;

4
sdk/android/src/jni/videoencoderwrapper.cc
View File

@ -165,10 +165,10 @@ VideoEncoderWrapper::ScalingSettings VideoEncoderWrapper::GetScalingSettings()
if (!isOn) if (!isOn)
return ScalingSettings::kOff; return ScalingSettings::kOff;
rtc::Optional<int> low = JavaToNativeOptionalInt( absl::optional<int> low = JavaToNativeOptionalInt(
jni, jni,
Java_VideoEncoderWrapper_getScalingSettingsLow(jni, j_scaling_settings)); Java_VideoEncoderWrapper_getScalingSettingsLow(jni, j_scaling_settings));
rtc::Optional<int> high = JavaToNativeOptionalInt( absl::optional<int> high = JavaToNativeOptionalInt(
jni, jni,
Java_VideoEncoderWrapper_getScalingSettingsHigh(jni, j_scaling_settings)); Java_VideoEncoderWrapper_getScalingSettingsHigh(jni, j_scaling_settings));

9
sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration.mm
View File

@ -171,8 +171,8 @@
// Generate non-default certificate. // Generate non-default certificate.
if (keyType != rtc::KT_DEFAULT) { if (keyType != rtc::KT_DEFAULT) {
rtc::scoped_refptr<rtc::RTCCertificate> certificate = rtc::scoped_refptr<rtc::RTCCertificate> certificate =
rtc::RTCCertificateGenerator::GenerateCertificate( rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(keyType),
rtc::KeyParams(keyType), rtc::Optional<uint64_t>()); absl::optional<uint64_t>());
if (!certificate) { if (!certificate) {
RTCLogError(@"Failed to generate certificate."); RTCLogError(@"Failed to generate certificate.");
return nullptr; return nullptr;
@ -184,14 +184,13 @@
nativeConfig->presume_writable_when_fully_relayed = nativeConfig->presume_writable_when_fully_relayed =
_shouldPresumeWritableWhenFullyRelayed ? true : false; _shouldPresumeWritableWhenFullyRelayed ? true : false;
if (_iceCheckMinInterval != nil) { if (_iceCheckMinInterval != nil) {
nativeConfig->ice_check_min_interval = nativeConfig->ice_check_min_interval = absl::optional<int>(_iceCheckMinInterval.intValue);
rtc::Optional<int>(_iceCheckMinInterval.intValue);
} }
if (_iceRegatherIntervalRange != nil) { if (_iceRegatherIntervalRange != nil) {
std::unique_ptr<rtc::IntervalRange> nativeIntervalRange( std::unique_ptr<rtc::IntervalRange> nativeIntervalRange(
_iceRegatherIntervalRange.nativeIntervalRange); _iceRegatherIntervalRange.nativeIntervalRange);
nativeConfig->ice_regather_interval_range = nativeConfig->ice_regather_interval_range =
rtc::Optional<rtc::IntervalRange>(*nativeIntervalRange); absl::optional<rtc::IntervalRange>(*nativeIntervalRange);
} }
nativeConfig->sdp_semantics = [[self class] nativeSdpSemanticsForSdpSemantics:_sdpSemantics]; nativeConfig->sdp_semantics = [[self class] nativeSdpSemanticsForSdpSemantics:_sdpSemantics];
if (_turnCustomizer) { if (_turnCustomizer) {

4
sdk/objc/Framework/Classes/PeerConnection/RTCH264ProfileLevelId.mm
View File

@ -31,7 +31,7 @@
if (self = [super init]) { if (self = [super init]) {
self.hexString = hexString; self.hexString = hexString;
rtc::Optional<webrtc::H264::ProfileLevelId> profile_level_id = absl::optional<webrtc::H264::ProfileLevelId> profile_level_id =
webrtc::H264::ParseProfileLevelId([hexString cStringUsingEncoding:NSUTF8StringEncoding]); webrtc::H264::ParseProfileLevelId([hexString cStringUsingEncoding:NSUTF8StringEncoding]);
if (profile_level_id.has_value()) { if (profile_level_id.has_value()) {
self.profile = static_cast<RTCH264Profile>(profile_level_id->profile); self.profile = static_cast<RTCH264Profile>(profile_level_id->profile);
@ -46,7 +46,7 @@
self.profile = profile; self.profile = profile;
self.level = level; self.level = level;
rtc::Optional<std::string> hex_string = absl::optional<std::string> hex_string =
webrtc::H264::ProfileLevelIdToString(webrtc::H264::ProfileLevelId( webrtc::H264::ProfileLevelIdToString(webrtc::H264::ProfileLevelId(
static_cast<webrtc::H264::Profile>(profile), static_cast<webrtc::H264::Level>(level))); static_cast<webrtc::H264::Profile>(profile), static_cast<webrtc::H264::Level>(level)));
self.hexString = self.hexString =

6
sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm
View File

@ -470,13 +470,13 @@ void PeerConnectionDelegateAdapter::OnAddTrack(
maxBitrateBps:(nullable NSNumber *)maxBitrateBps { maxBitrateBps:(nullable NSNumber *)maxBitrateBps {
webrtc::PeerConnectionInterface::BitrateParameters params; webrtc::PeerConnectionInterface::BitrateParameters params;
if (minBitrateBps != nil) { if (minBitrateBps != nil) {
params.min_bitrate_bps = rtc::Optional<int>(minBitrateBps.intValue); params.min_bitrate_bps = absl::optional<int>(minBitrateBps.intValue);
} }
if (currentBitrateBps != nil) { if (currentBitrateBps != nil) {
params.current_bitrate_bps = rtc::Optional<int>(currentBitrateBps.intValue); params.current_bitrate_bps = absl::optional<int>(currentBitrateBps.intValue);
} }
if (maxBitrateBps != nil) { if (maxBitrateBps != nil) {
params.max_bitrate_bps = rtc::Optional<int>(maxBitrateBps.intValue); params.max_bitrate_bps = absl::optional<int>(maxBitrateBps.intValue);
} }
return _peerConnection->SetBitrate(params).ok(); return _peerConnection->SetBitrate(params).ok();
} }

4
sdk/objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters.mm
View File

@ -93,10 +93,10 @@ const NSString * const kRTCH264CodecName = @(cricket::kH264CodecName);
RTC_NOTREACHED(); RTC_NOTREACHED();
} }
if (_clockRate != nil) { if (_clockRate != nil) {
parameters.clock_rate = rtc::Optional<int>(_clockRate.intValue); parameters.clock_rate = absl::optional<int>(_clockRate.intValue);
} }
if (_numChannels != nil) { if (_numChannels != nil) {
parameters.num_channels = rtc::Optional<int>(_numChannels.intValue); parameters.num_channels = absl::optional<int>(_numChannels.intValue);
} }
for (NSString *paramKey in _parameters.allKeys) { for (NSString *paramKey in _parameters.allKeys) {
std::string key = [NSString stdStringForString:paramKey]; std::string key = [NSString stdStringForString:paramKey];

6
sdk/objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters.mm
View File

@ -44,13 +44,13 @@
webrtc::RtpEncodingParameters parameters; webrtc::RtpEncodingParameters parameters;
parameters.active = _isActive; parameters.active = _isActive;
if (_maxBitrateBps != nil) { if (_maxBitrateBps != nil) {
parameters.max_bitrate_bps = rtc::Optional<int>(_maxBitrateBps.intValue); parameters.max_bitrate_bps = absl::optional<int>(_maxBitrateBps.intValue);
} }
if (_minBitrateBps != nil) { if (_minBitrateBps != nil) {
parameters.min_bitrate_bps = rtc::Optional<int>(_minBitrateBps.intValue); parameters.min_bitrate_bps = absl::optional<int>(_minBitrateBps.intValue);
} }
if (_ssrc != nil) { if (_ssrc != nil) {
parameters.ssrc = rtc::Optional<uint32_t>(_ssrc.unsignedLongValue); parameters.ssrc = absl::optional<uint32_t>(_ssrc.unsignedLongValue);
} }
return parameters; return parameters;
} }

4
sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec.mm
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@ -43,9 +43,9 @@ namespace {
using namespace webrtc::H264; using namespace webrtc::H264;
NSString *MaxSupportedLevelForProfile(Profile profile) { NSString *MaxSupportedLevelForProfile(Profile profile) {
const rtc::Optional<ProfileLevelId> profileLevelId = [UIDevice maxSupportedH264Profile]; const absl::optional<ProfileLevelId> profileLevelId = [UIDevice maxSupportedH264Profile];
if (profileLevelId && profileLevelId->profile >= profile) { if (profileLevelId && profileLevelId->profile >= profile) {
const rtc::Optional<std::string> profileString = const absl::optional<std::string> profileString =
ProfileLevelIdToString(ProfileLevelId(profile, profileLevelId->level)); ProfileLevelIdToString(ProfileLevelId(profile, profileLevelId->level));
if (profileString) { if (profileString) {
return [NSString stringForStdString:*profileString]; return [NSString stringForStdString:*profileString];

6
sdk/objc/Framework/Classes/Video/RTCDefaultShader.mm
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@ -20,7 +20,7 @@
#import "RTCShader.h" #import "RTCShader.h"
#import "WebRTC/RTCLogging.h" #import "WebRTC/RTCLogging.h"
#include "api/optional.h" #include "absl/types/optional.h"
static const int kYTextureUnit = 0; static const int kYTextureUnit = 0;
static const int kUTextureUnit = 1; static const int kUTextureUnit = 1;
@ -73,7 +73,7 @@ static const char kNV12FragmentShaderSource[] =
GLuint _vertexBuffer; GLuint _vertexBuffer;
GLuint _vertexArray; GLuint _vertexArray;
// Store current rotation and only upload new vertex data when rotation changes. // Store current rotation and only upload new vertex data when rotation changes.
rtc::Optional<RTCVideoRotation> _currentRotation; absl::optional<RTCVideoRotation> _currentRotation;
GLuint _i420Program; GLuint _i420Program;
GLuint _nv12Program; GLuint _nv12Program;
@ -144,7 +144,7 @@ static const char kNV12FragmentShaderSource[] =
#endif #endif
glBindBuffer(GL_ARRAY_BUFFER, _vertexBuffer); glBindBuffer(GL_ARRAY_BUFFER, _vertexBuffer);
if (!_currentRotation || rotation != *_currentRotation) { if (!_currentRotation || rotation != *_currentRotation) {
_currentRotation = rtc::Optional<RTCVideoRotation>(rotation); _currentRotation = absl::optional<RTCVideoRotation>(rotation);
RTCSetVertexData(*_currentRotation); RTCSetVertexData(*_currentRotation);
} }
return YES; return YES;

2
sdk/objc/Framework/Classes/Video/UIDevice+H264Profile.h
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@ -14,6 +14,6 @@
@interface UIDevice (H264Profile) @interface UIDevice (H264Profile)
+ (rtc::Optional<webrtc::H264::ProfileLevelId>)maxSupportedH264Profile; + (absl::optional<webrtc::H264::ProfileLevelId>)maxSupportedH264Profile;
@end @end

6
sdk/objc/Framework/Classes/Video/UIDevice+H264Profile.mm
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@ -85,7 +85,7 @@ constexpr SupportedH264Profile kH264MaxSupportedProfiles[] = {
{RTCDeviceTypeIPadPro10Inch, {kProfileHigh, kLevel4_2}}, // https://support.apple.com/kb/SP762 {RTCDeviceTypeIPadPro10Inch, {kProfileHigh, kLevel4_2}}, // https://support.apple.com/kb/SP762
}; };
rtc::Optional<ProfileLevelId> FindMaxSupportedProfileForDevice(RTCDeviceType deviceType) { absl::optional<ProfileLevelId> FindMaxSupportedProfileForDevice(RTCDeviceType deviceType) {
const auto* result = std::find_if(std::begin(kH264MaxSupportedProfiles), const auto* result = std::find_if(std::begin(kH264MaxSupportedProfiles),
std::end(kH264MaxSupportedProfiles), std::end(kH264MaxSupportedProfiles),
[deviceType](const SupportedH264Profile& supportedProfile) { [deviceType](const SupportedH264Profile& supportedProfile) {
@ -94,14 +94,14 @@ rtc::Optional<ProfileLevelId> FindMaxSupportedProfileForDevice(RTCDeviceType dev
if (result != std::end(kH264MaxSupportedProfiles)) { if (result != std::end(kH264MaxSupportedProfiles)) {
return result->profile; return result->profile;
} }
return rtc::nullopt; return absl::nullopt;
} }
} // namespace } // namespace
@implementation UIDevice (H264Profile) @implementation UIDevice (H264Profile)
+ (rtc::Optional<webrtc::H264::ProfileLevelId>)maxSupportedH264Profile { + (absl::optional<webrtc::H264::ProfileLevelId>)maxSupportedH264Profile {
return FindMaxSupportedProfileForDevice([self deviceType]); return FindMaxSupportedProfileForDevice([self deviceType]);
} }

2
sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
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@ -172,7 +172,7 @@ void compressionOutputCallback(void *encoder,
// returned. The user must initialize the encoder with a resolution and // returned. The user must initialize the encoder with a resolution and
// framerate conforming to the selected H264 level regardless. // framerate conforming to the selected H264 level regardless.
CFStringRef ExtractProfile(webrtc::SdpVideoFormat videoFormat) { CFStringRef ExtractProfile(webrtc::SdpVideoFormat videoFormat) {
const rtc::Optional<webrtc::H264::ProfileLevelId> profile_level_id = const absl::optional<webrtc::H264::ProfileLevelId> profile_level_id =
webrtc::H264::ParseSdpProfileLevelId(videoFormat.parameters); webrtc::H264::ParseSdpProfileLevelId(videoFormat.parameters);
RTC_DCHECK(profile_level_id); RTC_DCHECK(profile_level_id);
switch (profile_level_id->profile) { switch (profile_level_id->profile) {

2
sdk/objc/Framework/Native/src/objc_video_track_source.h
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@ -36,7 +36,7 @@ class ObjCVideoTrackSource : public rtc::AdaptedVideoTrackSource {
// Indicates that the encoder should denoise video before encoding it. // Indicates that the encoder should denoise video before encoding it.
// If it is not set, the default configuration is used which is different // If it is not set, the default configuration is used which is different
// depending on video codec. // depending on video codec.
rtc::Optional<bool> needs_denoising() const override { return false; } absl::optional<bool> needs_denoising() const override { return false; }
SourceState state() const override { return SourceState::kLive; } SourceState state() const override { return SourceState::kLive; }

2
system_wrappers/BUILD.gn
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@ -40,10 +40,10 @@ rtc_static_library("system_wrappers") {
":runtime_enabled_features_api", ":runtime_enabled_features_api",
"..:webrtc_common", "..:webrtc_common",
"../:typedefs", "../:typedefs",
"../api:optional",
"../modules:module_api_public", "../modules:module_api_public",
"../rtc_base:checks", "../rtc_base:checks",
"../rtc_base/synchronization:rw_lock_wrapper", "../rtc_base/synchronization:rw_lock_wrapper",
"//third_party/abseil-cpp/absl/types:optional",
] ]
if (is_posix || is_fuchsia) { if (is_posix || is_fuchsia) {

4
system_wrappers/include/rtp_to_ntp_estimator.h
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@ -13,7 +13,7 @@
#include <list> #include <list>
#include "api/optional.h" #include "absl/types/optional.h"
#include "modules/include/module_common_types_public.h" #include "modules/include/module_common_types_public.h"
#include "rtc_base/numerics/moving_median_filter.h" #include "rtc_base/numerics/moving_median_filter.h"
#include "system_wrappers/include/ntp_time.h" #include "system_wrappers/include/ntp_time.h"
@ -72,7 +72,7 @@ class RtpToNtpEstimator {
bool Estimate(int64_t rtp_timestamp, int64_t* rtp_timestamp_ms) const; bool Estimate(int64_t rtp_timestamp, int64_t* rtp_timestamp_ms) const;
// Returns estimated rtp to ntp linear transform parameters. // Returns estimated rtp to ntp linear transform parameters.
const rtc::Optional<Parameters> params() const; const absl::optional<Parameters> params() const;
static const int kMaxInvalidSamples = 3; static const int kMaxInvalidSamples = 3;

4
system_wrappers/source/rtp_to_ntp_estimator.cc
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@ -193,9 +193,9 @@ bool RtpToNtpEstimator::Estimate(int64_t rtp_timestamp,
return true; return true;
} }
const rtc::Optional<RtpToNtpEstimator::Parameters> RtpToNtpEstimator::params() const absl::optional<RtpToNtpEstimator::Parameters> RtpToNtpEstimator::params()
const { const {
rtc::Optional<Parameters> res; absl::optional<Parameters> res;
if (params_calculated_) { if (params_calculated_) {
res.emplace(smoothing_filter_.GetFilteredValue()); res.emplace(smoothing_filter_.GetFilteredValue());
} }