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Unified Diff: webrtc/modules/audio_coding/acm2/acm_receiver.h

Issue 1853183002: Change NetEq::GetPlayoutTimestamp to return an rtc::Optional<uint32_t> (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Adding back the old PlayoutTimestamp method, now DEPRECATED Created 4 years, 8 months ago
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Index: webrtc/modules/audio_coding/acm2/acm_receiver.h
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index 77eb563972377197c5d61c635248e04db209c4a1..6fec1ffdda1f965ed94f3cb8bbd6114755bfa2ed 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -195,11 +195,9 @@ class AcmReceiver {
//
int RemoveAllCodecs();
- //
- // Gets the RTP timestamp of the last sample delivered by GetAudio().
- // Returns true if the RTP timestamp is valid, otherwise false.
- //
- bool GetPlayoutTimestamp(uint32_t* timestamp);
+ // Returns the RTP timestamp for the last sample delivered by GetAudio().
+ // The return value will be empty if no valid timestamp is available.
+ rtc::Optional<uint32_t> GetPlayoutTimestamp();
//
// Get the audio codec associated with the last non-CNG/non-DTMF received

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