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