Change NetEq::GetPlayoutTimestamp to return an rtc::Optional<uint32_t>
This is in preparation for changes to when the playout timestamp is
valid.
BUG=webrtc:5669
Review URL: https://codereview.webrtc.org/1853183002
Cr-Commit-Position: refs/heads/master@{#12256}
diff --git a/webrtc/modules/audio_coding/test/delay_test.cc b/webrtc/modules/audio_coding/test/delay_test.cc
index 7288d50..8fa1fb1 100644
--- a/webrtc/modules/audio_coding/test/delay_test.cc
+++ b/webrtc/modules/audio_coding/test/delay_test.cc
@@ -180,7 +180,6 @@
int num_frames = 0;
int in_file_frames = 0;
- uint32_t playout_ts;
uint32_t received_ts;
double average_delay = 0;
double inst_delay_sec = 0;
@@ -209,10 +208,11 @@
out_file_b_.Write10MsData(
audio_frame.data_,
audio_frame.samples_per_channel_ * audio_frame.num_channels_);
- acm_b_->PlayoutTimestamp(&playout_ts);
received_ts = channel_a2b_->LastInTimestamp();
- inst_delay_sec = static_cast<uint32_t>(received_ts - playout_ts)
- / static_cast<double>(encoding_sample_rate_hz_);
+ rtc::Optional<uint32_t> playout_timestamp = acm_b_->PlayoutTimestamp();
+ ASSERT_TRUE(playout_timestamp);
+ inst_delay_sec = static_cast<uint32_t>(received_ts - *playout_timestamp) /
+ static_cast<double>(encoding_sample_rate_hz_);
if (num_frames > 10)
average_delay = 0.95 * average_delay + 0.05 * inst_delay_sec;