Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )

Reason for revert:
Breaks android bots.

Original issue's description:
> Make the new jitter buffer the default jitter buffer.
>
> This CL contains only the changes necessary to make the switch to the new jitter
> buffer, clean up will be done in follow up CLs.
>
> In this CL:
>  - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
>    new video jitter buffer the default one.
>  - Moved WebRTC.Video.KeyFramesReceivedInPermille and
>    WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.
>
> BUG=webrtc:5514
>
> Review-Url: https://codereview.webrtc.org/2627463004
> Cr-Commit-Position: refs/heads/master@{#16114}
> Committed: https://chromium.googlesource.com/external/webrtc/+/0f0763d86d5d4e7f27e8dece02560e39c6da97d6

TBR=stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2632123005
Cr-Commit-Position: refs/heads/master@{#16117}
diff --git a/webrtc/modules/video_coding/frame_buffer2.cc b/webrtc/modules/video_coding/frame_buffer2.cc
index db4928c..279c613 100644
--- a/webrtc/modules/video_coding/frame_buffer2.cc
+++ b/webrtc/modules/video_coding/frame_buffer2.cc
@@ -16,7 +16,6 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/modules/video_coding/include/video_coding_defines.h"
 #include "webrtc/modules/video_coding/jitter_estimator.h"
 #include "webrtc/modules/video_coding/timing.h"
 #include "webrtc/system_wrappers/include/clock.h"
@@ -35,8 +34,7 @@
 
 FrameBuffer::FrameBuffer(Clock* clock,
                          VCMJitterEstimator* jitter_estimator,
-                         VCMTiming* timing,
-                         VCMReceiveStatisticsCallback* stats_callback)
+                         VCMTiming* timing)
     : clock_(clock),
       new_countinuous_frame_event_(false, false),
       jitter_estimator_(jitter_estimator),
@@ -47,10 +45,11 @@
       num_frames_history_(0),
       num_frames_buffered_(0),
       stopped_(false),
-      protection_mode_(kProtectionNack),
-      stats_callback_(stats_callback) {}
+      protection_mode_(kProtectionNack) {}
 
-FrameBuffer::~FrameBuffer() {}
+FrameBuffer::~FrameBuffer() {
+  UpdateHistograms();
+}
 
 FrameBuffer::ReturnReason FrameBuffer::NextFrame(
     int64_t max_wait_time_ms,
@@ -163,8 +162,9 @@
   rtc::CritScope lock(&crit_);
   RTC_DCHECK(frame);
 
-  if (stats_callback_)
-    stats_callback_->OnCompleteFrame(frame->num_references == 0, frame->size());
+  ++num_total_frames_;
+  if (frame->num_references == 0)
+    ++num_key_frames_;
 
   FrameKey key(frame->picture_id, frame->spatial_layer);
   int last_continuous_picture_id =
@@ -365,22 +365,28 @@
 }
 
 void FrameBuffer::UpdateJitterDelay() {
-  if (!stats_callback_)
-    return;
+  int unused;
+  int delay;
+  timing_->GetTimings(&unused, &unused, &unused, &unused, &delay, &unused,
+                      &unused);
 
-  int decode_ms;
-  int max_decode_ms;
-  int current_delay_ms;
-  int target_delay_ms;
-  int jitter_buffer_ms;
-  int min_playout_delay_ms;
-  int render_delay_ms;
-  if (timing_->GetTimings(&decode_ms, &max_decode_ms, &current_delay_ms,
-                          &target_delay_ms, &jitter_buffer_ms,
-                          &min_playout_delay_ms, &render_delay_ms)) {
-    stats_callback_->OnFrameBufferTimingsUpdated(
-        decode_ms, max_decode_ms, current_delay_ms, target_delay_ms,
-        jitter_buffer_ms, min_playout_delay_ms, render_delay_ms);
+  accumulated_delay_ += delay;
+  ++accumulated_delay_samples_;
+}
+
+void FrameBuffer::UpdateHistograms() const {
+  rtc::CritScope lock(&crit_);
+  if (num_total_frames_ > 0) {
+    int key_frames_permille = (static_cast<float>(num_key_frames_) * 1000.0f /
+                                   static_cast<float>(num_total_frames_) +
+                               0.5f);
+    RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesReceivedInPermille",
+                              key_frames_permille);
+  }
+
+  if (accumulated_delay_samples_ > 0) {
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
+                               accumulated_delay_ / accumulated_delay_samples_);
   }
 }