Add PeerConnection option to configure minimum audio jitter buffer delay.

Note that this value will override the minimum delay that is used for audio/video sync.

Bug: webrtc:10053
Change-Id: Ia129f6c9ee9da5d00a3d955afaaa6e8f0c2bee33
Reviewed-on: https://webrtc-review.googlesource.com/c/112121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25805}

api/peerconnectioninterface.h

@@ -450,6 +450,9 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
     // if it falls behind.
     bool audio_jitter_buffer_fast_accelerate = false;
 
+    // The minimum delay in milliseconds for the audio jitter buffer.
+    int audio_jitter_buffer_min_delay_ms = 0;
+
     // Timeout in milliseconds before an ICE candidate pair is considered to be
     // "not receiving", after which a lower priority candidate pair may be
     // selected.