| Index: webrtc/voice_engine/channel.h
|
| diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
|
| index 07a0f789da8f22c907658f8e0ca838f8c7d975e5..2626df969b1456d9525408ac7f13cb40152b5bee 100644
|
| --- a/webrtc/voice_engine/channel.h
|
| +++ b/webrtc/voice_engine/channel.h
|
| @@ -15,6 +15,7 @@
|
|
|
| #include "webrtc/audio_sink.h"
|
| #include "webrtc/base/criticalsection.h"
|
| +#include "webrtc/base/optional.h"
|
| #include "webrtc/common_audio/resampler/include/push_resampler.h"
|
| #include "webrtc/common_types.h"
|
| #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
|
| @@ -501,7 +502,7 @@ class Channel
|
| RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_);
|
|
|
| // Timestamp of the audio pulled from NetEq.
|
| - uint32_t jitter_buffer_playout_timestamp_;
|
| + rtc::Optional<uint32_t> jitter_buffer_playout_timestamp_;
|
| uint32_t playout_timestamp_rtp_ GUARDED_BY(video_sync_lock_);
|
| uint32_t playout_timestamp_rtcp_;
|
| uint32_t playout_delay_ms_ GUARDED_BY(video_sync_lock_);
|
|
|