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Replacing rtc::TimeDelta with webrtc::TimeDelta.

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This removes the redundant type and replaces all usages. A slight change
in behavior is that we no longer get nanosecond resolution. This should
not matter since no current code requires nanosecond resolution.

Bug: webrtc:9155
Change-Id: I04334e08c686d95731621a6c8a7e40400d0ae3b2
Reviewed-on: https://webrtc-review.googlesource.com/71163
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23174}
This commit is contained in:
Sebastian Jansson
2018-05-08 14:52:22 +02:00
committed by Commit Bot
parent 5b2b692079
commit 5f83cf0c6d
24 changed files with 101 additions and 324 deletions
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12
modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
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@ -243,8 +243,7 @@ void TestSetPacketLossRate(const AudioEncoderOpusStates* states,
constexpr int64_t kSampleIntervalMs = 184198;
for (float loss : losses) {
states->encoder->OnReceivedUplinkPacketLossFraction(loss);
states->fake_clock->AdvanceTime(
rtc::TimeDelta::FromMilliseconds(kSampleIntervalMs));
states->fake_clock->AdvanceTime(TimeDelta::ms(kSampleIntervalMs));
EXPECT_FLOAT_EQ(expected_return, states->encoder->packet_loss_rate());
}
}
@ -376,8 +375,7 @@ TEST(AudioEncoderOpusTest,
states->encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_1);
EXPECT_FLOAT_EQ(0.01f, states->encoder->packet_loss_rate());
states->fake_clock->AdvanceTime(
rtc::TimeDelta::FromMilliseconds(kSecondSampleTimeMs));
states->fake_clock->AdvanceTime(TimeDelta::ms(kSecondSampleTimeMs));
states->encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_2);
// Now the output of packet loss fraction smoother should be
@ -564,8 +562,8 @@ TEST(AudioEncoderOpusTest, UpdateUplinkBandwidthInAudioNetworkAdaptor) {
// Repeat update uplink bandwidth tests.
for (int i = 0; i < 5; i++) {
// Don't update till it is time to update again.
states->fake_clock->AdvanceTime(rtc::TimeDelta::FromMilliseconds(
states->config.uplink_bandwidth_update_interval_ms - 1));
states->fake_clock->AdvanceTime(
TimeDelta::ms(states->config.uplink_bandwidth_update_interval_ms - 1));
states->encoder->Encode(
0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
@ -573,7 +571,7 @@ TEST(AudioEncoderOpusTest, UpdateUplinkBandwidthInAudioNetworkAdaptor) {
EXPECT_CALL(*states->mock_bitrate_smoother, GetAverage())
.WillOnce(Return(40000));
EXPECT_CALL(*states->mock_audio_network_adaptor, SetUplinkBandwidth(40000));
states->fake_clock->AdvanceTime(rtc::TimeDelta::FromMilliseconds(1));
states->fake_clock->AdvanceTime(TimeDelta::ms(1));
states->encoder->Encode(
0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
}