Optional: Use nullopt and implicit construction in /audio
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=solenberg@webrtc.org
Bug: None
Change-Id: I03562600978bdedb9dc93a34aeb0561c66f54aae
Reviewed-on: https://webrtc-review.googlesource.com/23617
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20731}diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index d8d81a4..8e42029 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -115,14 +115,15 @@
.WillByDefault(Return(std::vector<AudioCodecSpec>(
std::begin(kCodecSpecs), std::end(kCodecSpecs))));
ON_CALL(*factory.get(), QueryAudioEncoder(_))
- .WillByDefault(Invoke([](const SdpAudioFormat& format) {
- for (const auto& spec : kCodecSpecs) {
- if (format == spec.format) {
- return rtc::Optional<AudioCodecInfo>(spec.info);
- }
- }
- return rtc::Optional<AudioCodecInfo>();
- }));
+ .WillByDefault(Invoke(
+ [](const SdpAudioFormat& format) -> rtc::Optional<AudioCodecInfo> {
+ for (const auto& spec : kCodecSpecs) {
+ if (format == spec.format) {
+ return spec.info;
+ }
+ }
+ return rtc::nullopt;
+ }));
ON_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _))
.WillByDefault(Invoke([](int payload_type, const SdpAudioFormat& format,
std::unique_ptr<AudioEncoder>* return_value) {
@@ -168,8 +169,7 @@
// Use ISAC as default codec so as to prevent unnecessary |voice_engine_|
// calls from the default ctor behavior.
stream_config_.send_codec_spec =
- rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
- {kIsacPayloadType, kIsacFormat});
+ AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat);
stream_config_.voe_channel_id = kChannelId;
stream_config_.rtp.ssrc = kSsrc;
stream_config_.rtp.nack.rtp_history_ms = 200;
@@ -358,11 +358,10 @@
config.min_bitrate_bps = 12000;
config.max_bitrate_bps = 34000;
config.send_codec_spec =
- rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
- {kIsacPayloadType, kIsacFormat});
+ AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat);
config.send_codec_spec->nack_enabled = true;
config.send_codec_spec->transport_cc_enabled = false;
- config.send_codec_spec->cng_payload_type = rtc::Optional<int>(42);
+ config.send_codec_spec->cng_payload_type = 42;
config.encoder_factory = MockAudioEncoderFactory::CreateUnusedFactory();
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
@@ -383,7 +382,7 @@
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::Optional<RtpState>());
+ helper.rtcp_rtt_stats(), rtc::nullopt);
}
TEST(AudioSendStreamTest, SendTelephoneEvent) {
@@ -391,7 +390,7 @@
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::Optional<RtpState>());
+ helper.rtcp_rtt_stats(), rtc::nullopt);
helper.SetupMockForSendTelephoneEvent();
EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType,
kTelephoneEventPayloadFrequency, kTelephoneEventCode,
@@ -403,7 +402,7 @@
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::Optional<RtpState>());
+ helper.rtcp_rtt_stats(), rtc::nullopt);
EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true));
send_stream.SetMuted(true);
}
@@ -413,7 +412,7 @@
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::Optional<RtpState>());
+ helper.rtcp_rtt_stats(), rtc::nullopt);
}
TEST(AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) {
@@ -421,7 +420,7 @@
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::Optional<RtpState>());
+ helper.rtcp_rtt_stats(), rtc::nullopt);
}
TEST(AudioSendStreamTest, GetStats) {
@@ -429,7 +428,7 @@
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::Optional<RtpState>());
+ helper.rtcp_rtt_stats(), rtc::nullopt);
helper.SetupMockForGetStats();
AudioSendStream::Stats stats = send_stream.GetStats();
EXPECT_EQ(kSsrc, stats.local_ssrc);
@@ -461,12 +460,11 @@
ConfigHelper helper(false, true);
auto stream_config = helper.config();
stream_config.send_codec_spec =
- rtc::Optional<AudioSendStream::Config::SendCodecSpec>({0, kOpusFormat});
+ AudioSendStream::Config::SendCodecSpec(0, kOpusFormat);
const std::string kAnaConfigString = "abcde";
const std::string kAnaReconfigString = "12345";
- stream_config.audio_network_adaptor_config =
- rtc::Optional<std::string>(kAnaConfigString);
+ stream_config.audio_network_adaptor_config = kAnaConfigString;
EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _))
.WillOnce(Invoke([&kAnaConfigString, &kAnaReconfigString](
@@ -485,10 +483,9 @@
internal::AudioSendStream send_stream(
stream_config, helper.audio_state(), helper.worker_queue(),
helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::Optional<RtpState>());
+ helper.rtcp_rtt_stats(), rtc::nullopt);
- stream_config.audio_network_adaptor_config =
- rtc::Optional<std::string>(kAnaReconfigString);
+ stream_config.audio_network_adaptor_config = kAnaReconfigString;
helper.SetupMockForModifyEncoder();
send_stream.Reconfigure(stream_config);
@@ -500,8 +497,8 @@
ConfigHelper helper(false, false);
auto stream_config = helper.config();
stream_config.send_codec_spec =
- rtc::Optional<AudioSendStream::Config::SendCodecSpec>({9, kG722Format});
- stream_config.send_codec_spec->cng_payload_type = rtc::Optional<int>(105);
+ AudioSendStream::Config::SendCodecSpec(9, kG722Format);
+ stream_config.send_codec_spec->cng_payload_type = 105;
using ::testing::Invoke;
std::unique_ptr<AudioEncoder> stolen_encoder;
EXPECT_CALL(*helper.channel_proxy(), SetEncoderForMock(_, _))
@@ -515,7 +512,7 @@
internal::AudioSendStream send_stream(
stream_config, helper.audio_state(), helper.worker_queue(),
helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::Optional<RtpState>());
+ helper.rtcp_rtt_stats(), rtc::nullopt);
// We cannot truly determine if the encoder created is an AudioEncoderCng. It
// is the only reasonable implementation that will return something from
@@ -529,7 +526,7 @@
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::Optional<RtpState>());
+ helper.rtcp_rtt_stats(), rtc::nullopt);
EXPECT_CALL(*helper.channel_proxy(),
SetBitrate(helper.config().max_bitrate_bps, _));
send_stream.OnBitrateUpdated(helper.config().max_bitrate_bps + 5000, 0.0, 50,
@@ -541,7 +538,7 @@
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::Optional<RtpState>());
+ helper.rtcp_rtt_stats(), rtc::nullopt);
EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000));
send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000);
}
@@ -559,12 +556,12 @@
auto stream_config = helper.config();
stream_config.send_codec_spec =
- rtc::Optional<AudioSendStream::Config::SendCodecSpec>({9, kG722Format});
- stream_config.send_codec_spec->cng_payload_type = rtc::Optional<int>(105);
+ AudioSendStream::Config::SendCodecSpec(9, kG722Format);
+ stream_config.send_codec_spec->cng_payload_type = 105;
internal::AudioSendStream send_stream(
stream_config, helper.audio_state(), helper.worker_queue(),
helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::Optional<RtpState>());
+ helper.rtcp_rtt_stats(), rtc::nullopt);
send_stream.Reconfigure(stream_config);
}
@@ -573,7 +570,7 @@
internal::AudioSendStream send_stream(
helper.config(), helper.audio_state(), helper.worker_queue(),
helper.transport(), helper.bitrate_allocator(), helper.event_log(),
- helper.rtcp_rtt_stats(), rtc::Optional<RtpState>());
+ helper.rtcp_rtt_stats(), rtc::nullopt);
auto new_config = helper.config();
ConfigHelper::AddBweToConfig(&new_config);
EXPECT_CALL(*helper.channel_proxy(),