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Replace rtc::Optional with absl::optional in pc

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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 with parameter 'pc'

find $@ -type f \( -name \*.h -o -name \*.cc \) \
-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: Ide3b9eb32df7f25991f898ac58fcb119c9f8ae12
Reviewed-on: https://webrtc-review.googlesource.com/84181
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23669}
This commit is contained in:
Danil Chapovalov
2018-06-19 16:47:43 +02:00
committed by Commit Bot
parent 751a817044
commit 66cadcc6b9
52 changed files with 196 additions and 191 deletions
Download Patch File Download Diff File Expand all files Collapse all files

20
pc/peerconnectionendtoend_unittest.cc
View File

@ -270,7 +270,7 @@ CreateForwardingMockDecoderFactory(
.WillRepeatedly(
Invoke([real_decoder_factory](
const webrtc::SdpAudioFormat& format,
rtc::Optional<webrtc::AudioCodecPairId> codec_pair_id,
absl::optional<webrtc::AudioCodecPairId> codec_pair_id,
std::unique_ptr<webrtc::AudioDecoder>* return_value) {
auto real_decoder =
real_decoder_factory->MakeAudioDecoder(format, codec_pair_id);
@ -284,7 +284,7 @@ CreateForwardingMockDecoderFactory(
struct AudioEncoderUnicornSparklesRainbow {
using Config = webrtc::AudioEncoderL16::Config;
static rtc::Optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
static absl::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
if (STR_CASE_CMP(format.name.c_str(), "UnicornSparklesRainbow") == 0) {
const webrtc::SdpAudioFormat::Parameters expected_params = {
{"num_horns", "1"}};
@ -293,7 +293,7 @@ struct AudioEncoderUnicornSparklesRainbow {
format.name = "L16";
return webrtc::AudioEncoderL16::SdpToConfig(format);
} else {
return rtc::nullopt;
return absl::nullopt;
}
}
static void AppendSupportedEncoders(
@ -313,7 +313,7 @@ struct AudioEncoderUnicornSparklesRainbow {
static std::unique_ptr<webrtc::AudioEncoder> MakeAudioEncoder(
const Config& config,
int payload_type,
rtc::Optional<webrtc::AudioCodecPairId> codec_pair_id = rtc::nullopt) {
absl::optional<webrtc::AudioCodecPairId> codec_pair_id = absl::nullopt) {
return webrtc::AudioEncoderL16::MakeAudioEncoder(config, payload_type,
codec_pair_id);
}
@ -321,7 +321,7 @@ struct AudioEncoderUnicornSparklesRainbow {
struct AudioDecoderUnicornSparklesRainbow {
using Config = webrtc::AudioDecoderL16::Config;
static rtc::Optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
static absl::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
if (STR_CASE_CMP(format.name.c_str(), "UnicornSparklesRainbow") == 0) {
const webrtc::SdpAudioFormat::Parameters expected_params = {
{"num_horns", "1"}};
@ -330,7 +330,7 @@ struct AudioDecoderUnicornSparklesRainbow {
format.name = "L16";
return webrtc::AudioDecoderL16::SdpToConfig(format);
} else {
return rtc::nullopt;
return absl::nullopt;
}
}
static void AppendSupportedDecoders(
@ -346,7 +346,7 @@ struct AudioDecoderUnicornSparklesRainbow {
}
static std::unique_ptr<webrtc::AudioDecoder> MakeAudioDecoder(
const Config& config,
rtc::Optional<webrtc::AudioCodecPairId> codec_pair_id = rtc::nullopt) {
absl::optional<webrtc::AudioCodecPairId> codec_pair_id = absl::nullopt) {
return webrtc::AudioDecoderL16::MakeAudioDecoder(config, codec_pair_id);
}
};
@ -392,14 +392,14 @@ TEST_P(PeerConnectionEndToEndTest, CallWithCustomCodec) {
std::vector<webrtc::AudioCodecSpec> GetSupportedEncoders() override {
return fact_->GetSupportedEncoders();
}
rtc::Optional<webrtc::AudioCodecInfo> QueryAudioEncoder(
absl::optional<webrtc::AudioCodecInfo> QueryAudioEncoder(
const webrtc::SdpAudioFormat& format) override {
return fact_->QueryAudioEncoder(format);
}
std::unique_ptr<webrtc::AudioEncoder> MakeAudioEncoder(
int payload_type,
const webrtc::SdpAudioFormat& format,
rtc::Optional<webrtc::AudioCodecPairId> codec_pair_id) override {
absl::optional<webrtc::AudioCodecPairId> codec_pair_id) override {
EXPECT_TRUE(codec_pair_id.has_value());
codec_ids_->push_back(*codec_pair_id);
return fact_->MakeAudioEncoder(payload_type, format, codec_pair_id);
@ -424,7 +424,7 @@ TEST_P(PeerConnectionEndToEndTest, CallWithCustomCodec) {
}
std::unique_ptr<webrtc::AudioDecoder> MakeAudioDecoder(
const webrtc::SdpAudioFormat& format,
rtc::Optional<webrtc::AudioCodecPairId> codec_pair_id) override {
absl::optional<webrtc::AudioCodecPairId> codec_pair_id) override {
EXPECT_TRUE(codec_pair_id.has_value());
codec_ids_->push_back(*codec_pair_id);
return fact_->MakeAudioDecoder(format, codec_pair_id);