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Change NetEq::GetPlayoutTimestamp to return an rtc::Optional<uint32_t>

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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}
This commit is contained in:
henrik.lundin
2016-04-06 01:39:22 -07:00
committed by Commit bot
parent 05255b0e8a
commit 9a410dd082
13 changed files with 77 additions and 58 deletions
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8
webrtc/modules/audio_coding/test/delay_test.cc
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@ -180,7 +180,6 @@ class DelayTest {
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 @@ class DelayTest {
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;