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This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
109 lines
3.4 KiB
C++
109 lines
3.4 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/aec3/api_call_jitter_metrics.h"
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#include "modules/audio_processing/aec3/aec3_common.h"
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#include "test/gtest.h"
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namespace webrtc {
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// Verify constant jitter.
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TEST(ApiCallJitterMetrics, ConstantJitter) {
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for (int jitter = 1; jitter < 20; ++jitter) {
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ApiCallJitterMetrics metrics;
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for (size_t k = 0; k < 30 * kNumBlocksPerSecond; ++k) {
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for (int j = 0; j < jitter; ++j) {
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metrics.ReportRenderCall();
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}
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for (int j = 0; j < jitter; ++j) {
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metrics.ReportCaptureCall();
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if (metrics.WillReportMetricsAtNextCapture()) {
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EXPECT_EQ(jitter, metrics.render_jitter().min());
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EXPECT_EQ(jitter, metrics.render_jitter().max());
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EXPECT_EQ(jitter, metrics.capture_jitter().min());
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EXPECT_EQ(jitter, metrics.capture_jitter().max());
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}
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}
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}
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}
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}
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// Verify peaky jitter for the render.
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TEST(ApiCallJitterMetrics, JitterPeakRender) {
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constexpr int kMinJitter = 2;
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constexpr int kJitterPeak = 10;
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constexpr int kPeakInterval = 100;
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ApiCallJitterMetrics metrics;
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int render_surplus = 0;
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for (size_t k = 0; k < 30 * kNumBlocksPerSecond; ++k) {
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const int num_render_calls =
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k % kPeakInterval == 0 ? kJitterPeak : kMinJitter;
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for (int j = 0; j < num_render_calls; ++j) {
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metrics.ReportRenderCall();
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++render_surplus;
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}
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ASSERT_LE(kMinJitter, render_surplus);
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const int num_capture_calls =
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render_surplus == kMinJitter ? kMinJitter : kMinJitter + 1;
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for (int j = 0; j < num_capture_calls; ++j) {
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metrics.ReportCaptureCall();
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if (metrics.WillReportMetricsAtNextCapture()) {
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EXPECT_EQ(kMinJitter, metrics.render_jitter().min());
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EXPECT_EQ(kJitterPeak, metrics.render_jitter().max());
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EXPECT_EQ(kMinJitter, metrics.capture_jitter().min());
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EXPECT_EQ(kMinJitter + 1, metrics.capture_jitter().max());
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}
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--render_surplus;
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}
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}
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}
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// Verify peaky jitter for the capture.
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TEST(ApiCallJitterMetrics, JitterPeakCapture) {
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constexpr int kMinJitter = 2;
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constexpr int kJitterPeak = 10;
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constexpr int kPeakInterval = 100;
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ApiCallJitterMetrics metrics;
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int capture_surplus = kMinJitter;
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for (size_t k = 0; k < 30 * kNumBlocksPerSecond; ++k) {
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ASSERT_LE(kMinJitter, capture_surplus);
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const int num_render_calls =
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capture_surplus == kMinJitter ? kMinJitter : kMinJitter + 1;
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for (int j = 0; j < num_render_calls; ++j) {
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metrics.ReportRenderCall();
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--capture_surplus;
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}
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const int num_capture_calls =
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k % kPeakInterval == 0 ? kJitterPeak : kMinJitter;
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for (int j = 0; j < num_capture_calls; ++j) {
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metrics.ReportCaptureCall();
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if (metrics.WillReportMetricsAtNextCapture()) {
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EXPECT_EQ(kMinJitter, metrics.render_jitter().min());
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EXPECT_EQ(kMinJitter + 1, metrics.render_jitter().max());
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EXPECT_EQ(kMinJitter, metrics.capture_jitter().min());
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EXPECT_EQ(kJitterPeak, metrics.capture_jitter().max());
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}
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++capture_surplus;
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}
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}
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}
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} // namespace webrtc
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