Delete AudioCodingModule::LeastRequiredDelayMs and related NetEq code.

Bug: None
Change-Id: I2f68502d19415899b3694f7bf5da523da831b223
Reviewed-on: https://webrtc-review.googlesource.com/95640
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24439}
This commit is contained in:
Niels Möller
2018-08-23 08:40:41 +02:00
committed by Commit Bot
parent 8d92e8d323
commit 18f1adc0da
11 changed files with 2 additions and 123 deletions

View File

@ -602,15 +602,6 @@ class AudioCodingModule {
//
virtual int SetMaximumPlayoutDelay(int time_ms) = 0;
// TODO(kwiberg): Consider if this is needed anymore, now that voe::Channel
// doesn't use it.
// The shortest latency, in milliseconds, required by jitter buffer. This
// is computed based on inter-arrival times and playout mode of NetEq. The
// actual delay is the maximum of least-required-delay and the minimum-delay
// specified by SetMinumumPlayoutDelay() API.
//
virtual int LeastRequiredDelayMs() const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t PlayoutTimestamp()
// The send timestamp of an RTP packet is associated with the decoded