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/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index 5649f07..925e99c 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -196,9 +196,10 @@
// |GetPlayoutTimestamp|, which is the timestamp of the last sample of
// |audio_frame|.
// TODO(henrik.lundin) Move setting of audio_frame->timestamp_ inside NetEq.
- uint32_t playout_timestamp = 0;
- if (GetPlayoutTimestamp(&playout_timestamp)) {
- audio_frame->timestamp_ = playout_timestamp -
+ rtc::Optional<uint32_t> playout_timestamp = GetPlayoutTimestamp();
+ if (playout_timestamp) {
+ audio_frame->timestamp_ =
+ *playout_timestamp -
static_cast<uint32_t>(audio_frame->samples_per_channel_);
} else {
// Remain 0 until we have a valid |playout_timestamp|.
@@ -318,8 +319,8 @@
return 0;
}
-bool AcmReceiver::GetPlayoutTimestamp(uint32_t* timestamp) {
- return neteq_->GetPlayoutTimestamp(timestamp);
+rtc::Optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
+ return neteq_->GetPlayoutTimestamp();
}
int AcmReceiver::LastAudioCodec(CodecInst* codec) const {