Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1045)

Unified Diff: webrtc/modules/video_coding/media_optimization_unittest.cc

Issue 2119503002: Remove all old suspension logic. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 6 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/modules/video_coding/media_optimization.cc ('k') | webrtc/modules/video_coding/video_coding_impl.h » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/modules/video_coding/media_optimization_unittest.cc
diff --git a/webrtc/modules/video_coding/media_optimization_unittest.cc b/webrtc/modules/video_coding/media_optimization_unittest.cc
deleted file mode 100644
index 2263099f238b119e2a69e768a1ae0bc5361c5b26..0000000000000000000000000000000000000000
--- a/webrtc/modules/video_coding/media_optimization_unittest.cc
+++ /dev/null
@@ -1,111 +0,0 @@
-/*
- * Copyright (c) 2013 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 "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/video_coding/media_optimization.h"
-#include "webrtc/system_wrappers/include/clock.h"
-
-namespace webrtc {
-namespace media_optimization {
-
-class TestMediaOptimization : public ::testing::Test {
- protected:
- enum {
- kSampleRate = 90000 // RTP timestamps per second.
- };
-
- // Note: simulated clock starts at 1 seconds, since parts of webrtc use 0 as
- // a special case (e.g. frame rate in media optimization).
- TestMediaOptimization()
- : clock_(1000),
- media_opt_(&clock_),
- frame_time_ms_(33),
- next_timestamp_(0) {}
-
- // This method mimics what happens in VideoSender::AddVideoFrame.
- void AddFrameAndAdvanceTime(uint32_t bitrate_bps, bool expect_frame_drop) {
- bool frame_dropped = media_opt_.DropFrame();
- EXPECT_EQ(expect_frame_drop, frame_dropped);
- if (!frame_dropped) {
- size_t bytes_per_frame = bitrate_bps * frame_time_ms_ / (8 * 1000);
- EncodedImage encoded_image;
- encoded_image._length = bytes_per_frame;
- encoded_image._timeStamp = next_timestamp_;
- encoded_image._frameType = kVideoFrameKey;
- ASSERT_EQ(VCM_OK, media_opt_.UpdateWithEncodedData(encoded_image));
- }
- next_timestamp_ += frame_time_ms_ * kSampleRate / 1000;
- clock_.AdvanceTimeMilliseconds(frame_time_ms_);
- }
-
- SimulatedClock clock_;
- MediaOptimization media_opt_;
- int frame_time_ms_;
- uint32_t next_timestamp_;
-};
-
-TEST_F(TestMediaOptimization, VerifyMuting) {
- // Enable video suspension with these limits.
- // Suspend the video when the rate is below 50 kbps and resume when it gets
- // above 50 + 10 kbps again.
- const uint32_t kThresholdBps = 50000;
- const uint32_t kWindowBps = 10000;
- media_opt_.SuspendBelowMinBitrate(kThresholdBps, kWindowBps);
-
- // The video should not be suspended from the start.
- EXPECT_FALSE(media_opt_.IsVideoSuspended());
-
- uint32_t target_bitrate_kbps = 100;
- media_opt_.SetTargetRates(target_bitrate_kbps * 1000,
- 0, // Lossrate.
- 100); // RTT in ms.
- media_opt_.EnableFrameDropper(true);
- for (int time = 0; time < 2000; time += frame_time_ms_) {
- ASSERT_NO_FATAL_FAILURE(AddFrameAndAdvanceTime(target_bitrate_kbps, false));
- }
-
- // Set the target rate below the limit for muting.
- media_opt_.SetTargetRates(kThresholdBps - 1000,
- 0, // Lossrate.
- 100); // RTT in ms.
- // Expect the muter to engage immediately and stay muted.
- // Test during 2 seconds.
- for (int time = 0; time < 2000; time += frame_time_ms_) {
- EXPECT_TRUE(media_opt_.IsVideoSuspended());
- ASSERT_NO_FATAL_FAILURE(AddFrameAndAdvanceTime(target_bitrate_kbps, true));
- }
-
- // Set the target above the limit for muting, but not above the
- // limit + window.
- media_opt_.SetTargetRates(kThresholdBps + 1000,
- 0, // Lossrate.
- 100); // RTT in ms.
- // Expect the muter to stay muted.
- // Test during 2 seconds.
- for (int time = 0; time < 2000; time += frame_time_ms_) {
- EXPECT_TRUE(media_opt_.IsVideoSuspended());
- ASSERT_NO_FATAL_FAILURE(AddFrameAndAdvanceTime(target_bitrate_kbps, true));
- }
-
- // Set the target above limit + window.
- media_opt_.SetTargetRates(kThresholdBps + kWindowBps + 1000,
- 0, // Lossrate.
- 100); // RTT in ms.
- // Expect the muter to disengage immediately.
- // Test during 2 seconds.
- for (int time = 0; time < 2000; time += frame_time_ms_) {
- EXPECT_FALSE(media_opt_.IsVideoSuspended());
- ASSERT_NO_FATAL_FAILURE(
- AddFrameAndAdvanceTime((kThresholdBps + kWindowBps) / 1000, false));
- }
-}
-
-} // namespace media_optimization
-} // namespace webrtc
« no previous file with comments | « webrtc/modules/video_coding/media_optimization.cc ('k') | webrtc/modules/video_coding/video_coding_impl.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698