Remove SetLatency/GetLatency from MediaSourceInterface API level
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}diff --git a/api/rtp_receiver_interface.h b/api/rtp_receiver_interface.h
index d1ef137..3934215 100644
--- a/api/rtp_receiver_interface.h
+++ b/api/rtp_receiver_interface.h
@@ -131,10 +131,9 @@
// 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