Partial merge of WebRTC

This is a partial merge. This is needed since we do not accept changes
to OWNER files from the chrome domain, so those cannot come in a single
merge.

Change-Id: I37473f53ec79e422e8b77761a5859ebcd73f2e3e

1 files changed, 12 insertions, 11 deletions

diff --git a/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h b/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h
index f5ec820dd5..c830686359 100644
--- a/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h
+++ b/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h
@@ -35,8 +35,7 @@ namespace webrtc {
 //
 class AbsoluteCaptureTimeInterpolator {
  public:
-  static constexpr TimeDelta kInterpolationMaxInterval =
-      TimeDelta::Millis(5000);
+  static constexpr TimeDelta kInterpolationMaxInterval = TimeDelta::Seconds(5);
 
   explicit AbsoluteCaptureTimeInterpolator(Clock* clock);
 
@@ -49,7 +48,7 @@ class AbsoluteCaptureTimeInterpolator {
   absl::optional<AbsoluteCaptureTime> OnReceivePacket(
       uint32_t source,
       uint32_t rtp_timestamp,
-      uint32_t rtp_clock_frequency,
+      int rtp_clock_frequency_hz,
       const absl::optional<AbsoluteCaptureTime>& received_extension);
 
  private:
@@ -57,29 +56,31 @@ class AbsoluteCaptureTimeInterpolator {
 
   static uint64_t InterpolateAbsoluteCaptureTimestamp(
       uint32_t rtp_timestamp,
-      uint32_t rtp_clock_frequency,
+      int rtp_clock_frequency_hz,
       uint32_t last_rtp_timestamp,
       uint64_t last_absolute_capture_timestamp);
 
   bool ShouldInterpolateExtension(Timestamp receive_time,
                                   uint32_t source,
                                   uint32_t rtp_timestamp,
-                                  uint32_t rtp_clock_frequency) const
+                                  int rtp_clock_frequency_hz) const
       RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
 
   Clock* const clock_;
 
   Mutex mutex_;
 
-  Timestamp last_receive_time_ RTC_GUARDED_BY(mutex_);
+  // Time of the last received header extension eligible for interpolation,
+  // MinusInfinity() if no extension was received, or last received one is
+  // not eligible for interpolation.
+  Timestamp last_receive_time_ RTC_GUARDED_BY(mutex_) =
+      Timestamp::MinusInfinity();
 
   uint32_t last_source_ RTC_GUARDED_BY(mutex_);
   uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(mutex_);
-  uint32_t last_rtp_clock_frequency_ RTC_GUARDED_BY(mutex_);
-  uint64_t last_absolute_capture_timestamp_ RTC_GUARDED_BY(mutex_);
-  absl::optional<int64_t> last_estimated_capture_clock_offset_
-      RTC_GUARDED_BY(mutex_);
-};  // AbsoluteCaptureTimeInterpolator
+  int last_rtp_clock_frequency_hz_ RTC_GUARDED_BY(mutex_);
+  AbsoluteCaptureTime last_received_extension_ RTC_GUARDED_BY(mutex_);
+};
 
 }  // namespace webrtc