commit 791294a647bfba8ebd26821a78020a2bb2f82b9b
author Danil Chapovalov <danilchap@webrtc.org> Thu Jul 07 07:18:01 2022 +0000
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Thu Jul 07 07:19:44 2022 +0000
tree 20af4418ad1e40bcb9b3f00acd1e013d7774ed06
parent 3f207658da56e850c711b4afd51b825c06be71cf

Revert "Fix overflow due to rounding in AbsoluteSendTime::To24Bits"

This reverts commit a17651f7d8748905d902eedf34471a0c227ca789.

Reason for revert: triggers failure in downstream test

Original change's description:
> Fix overflow due to rounding in AbsoluteSendTime::To24Bits
>
> Actual rounding is not important for this time as long it is consistent
> during the call: only difference between two absolute send time matter
> Rounding down avoids producing 1 < 24 when value is close to the wrap around boundary.
>
> Bug: None
> Change-Id: Ibbf5bae21bc37eccdc5d4c130a59796ee5108017
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268001
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37468}

Bug: None
Change-Id: I90a9c1b174b918b7ede58c3bbdb879b1b67da7b2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268120
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37473}

modules/rtp_rtcp/source/rtp_header_extensions.h

diff --git a/modules/rtp_rtcp/source/rtp_header_extensions.h b/modules/rtp_rtcp/source/rtp_header_extensions.h
index d80e0da..62b1929 100644
--- a/modules/rtp_rtcp/source/rtp_header_extensions.h
+++ b/modules/rtp_rtcp/source/rtp_header_extensions.h
@@ -26,6 +26,7 @@
 #include "api/video/video_rotation.h"
 #include "api/video/video_timing.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/numerics/divide_round.h"
 
 namespace webrtc {
 
@@ -44,7 +45,7 @@
 
   static constexpr uint32_t To24Bits(Timestamp time) {
     int64_t time_us = time.us() % (int64_t{1 << 6} * 1'000'000);
-    int64_t time6x18 = (time_us << 18) / 1'000'000;
+    int64_t time6x18 = DivideRoundToNearest(time_us << 18, 1'000'000);
     RTC_DCHECK_GE(time6x18, 0);
     RTC_DCHECK_LT(time6x18, 1 << 24);
     return static_cast<uint32_t>(time6x18);