webrtc/modules
Danil Chapovalov 791294a647 Revert "Fix overflow due to rounding in AbsoluteSendTime::To24Bits"
This reverts commit a17651f7d8.

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}
2022-07-07 07:19:44 +00:00
..
async_audio_processing Remove dependency on rtc_base_approved from most targets 2022-04-25 12:15:30 +00:00
audio_coding Implement RTCInboundRTPStreamStats.JitterBufferTargetDelay 2022-07-05 11:34:53 +00:00
audio_device Update/delete old TODOs 2022-07-06 07:49:04 +00:00
audio_mixer Move rtc::make_ref_counted to api/ 2022-06-15 09:47:38 +00:00
audio_processing AgcManagerDirect: stop enforcing min mic level override with 0 level 2022-07-06 09:50:43 +00:00
congestion_controller Add absl::string_view overload for RtcEventLogOutput::Write 2022-07-05 10:47:47 +00:00
desktop_capture Reland "Wait for frames to arrive in WgcCapturer instead of returning nothing." 2022-07-06 20:28:26 +00:00
include Delete ProcessThread and related Module interface 2022-07-04 10:20:35 +00:00
pacing Migrate pacing and video_coding to absl::AnyInvocable based TaskQueueBase interface 2022-07-06 15:46:04 +00:00
remote_bitrate_estimator Cleanup RemoteBitrateEstimate::LatestEstimate function 2022-07-01 13:05:05 +00:00
rtp_rtcp Revert "Fix overflow due to rounding in AbsoluteSendTime::To24Bits" 2022-07-07 07:19:44 +00:00
third_party Update portaudio to the latest 2022-05-13 09:01:34 +00:00
utility Delete ProcessThread and related Module interface 2022-07-04 10:20:35 +00:00
video_capture Update/delete old TODOs 2022-07-06 07:49:04 +00:00
video_coding Migrate pacing and video_coding to absl::AnyInvocable based TaskQueueBase interface 2022-07-06 15:46:04 +00:00
video_processing Delete ProcessThread and related Module interface 2022-07-04 10:20:35 +00:00
BUILD.gn Delete ProcessThread and related Module interface 2022-07-04 10:20:35 +00:00
module_common_types_unittest.cc