zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Remove SetLatency/GetLatency from MediaSourceInterface API level

Browse Source 
Bug: webrtc:10287
Change-Id: I74fad31db98b75791085688438064f9510b0b6fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133165
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27692}
This commit is contained in:
Ruslan Burakov
2019-04-18 17:49:49 +02:00
committed by Commit Bot
parent 81687b370f
commit 428dcb2517
19 changed files with 173 additions and 333 deletions
Download Patch File Download Diff File Expand all files Collapse all files

5
api/rtp_receiver_interface.h
View File

@ -131,10 +131,9 @@ class RtpReceiverInterface : public rtc::RefCountInterface {
// Sets the jitter buffer minimum delay until media playout. Actual observed
// delay may differ depending on the congestion control. |delay_seconds| is a
// positive value including 0.0 measured in seconds. |nullopt| means default
// value must be used. TODO(kuddai): remove the default implmenetation once
// the subclasses in Chromium implement this.
// value must be used.
virtual void SetJitterBufferMinimumDelay(
absl::optional<double> delay_seconds);
absl::optional<double> delay_seconds) = 0;
// TODO(zhihuang): Remove the default implementation once the subclasses
// implement this. Currently, the only relevant subclass is the