Sergey Silkin 18bc3e19c4 Revert "Updated analysis in videoprocessor."
This reverts commit 1880c7162bd3637c433f9421c798808cd6eacaf7.

Reason for revert: breaks internal tests

Original change's description:
> Updated analysis in videoprocessor.
> 
> - Run analysis after all frames are processed. Before part of it was
> done at bitrate change points;
> - Analysis is done for whole stream as well as for each rate update
> interval;
> - Changed units from number of frames to time units for some metrics
> and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to
> 'time to reach target bitrate, sec';
> - Changed data type of FrameStatistic::max_nalu_length (renamed to
> max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to
> use such advanced data type in such low level data structure.
> 
> Bug: webrtc:8524
> Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f
> Reviewed-on: https://webrtc-review.googlesource.com/31901
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21653}

TBR=brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,ssilkin@webrtc.org

Change-Id: Id0b7d387bbba02e71637b229aeed6f6cf012af46
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8524
Reviewed-on: https://webrtc-review.googlesource.com/40220
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21656}
2018-01-17 13:16:07 +00:00

44 lines
1.1 KiB
C++

/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/video_coding/codecs/test/stats.h"
#include "test/gtest.h"
namespace webrtc {
namespace test {
TEST(StatsTest, TestEmptyObject) {
Stats stats;
stats.PrintSummary(); // Should not crash.
}
TEST(StatsTest, AddSingleFrame) {
Stats stats;
FrameStatistic* frame_stat = stats.AddFrame();
EXPECT_EQ(0, frame_stat->frame_number);
EXPECT_EQ(1u, stats.size());
}
TEST(StatsTest, AddMultipleFrames) {
Stats stats;
const int kNumFrames = 1000;
for (int i = 0; i < kNumFrames; ++i) {
FrameStatistic* frame_stat = stats.AddFrame();
EXPECT_EQ(i, frame_stat->frame_number);
}
EXPECT_EQ(kNumFrames, static_cast<int>(stats.size()));
stats.PrintSummary(); // Should not crash.
}
} // namespace test
} // namespace webrtc