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authorChih-hung Hsieh <chh@google.com>2015-12-01 17:07:48 +0000
committerandroid-build-merger <android-build-merger@google.com>2015-12-01 17:07:48 +0000
commita4acd9d6bc9b3b033d7d274316e75ee067df8d20 (patch)
tree672a185b294789cf991f385c3e395dd63bea9063 /webrtc/video_engine/vie_remb_unittest.cc
parent3681b90ba4fe7a27232dd3e27897d5d7ed9d651c (diff)
parentfe8b4a657979b49e1701bd92f6d5814a99e0b2be (diff)
downloadwebrtc-a4acd9d6bc9b3b033d7d274316e75ee067df8d20.tar.gz
Merge changes I7bbf776e,I1b827825
am: fe8b4a6579 * commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits) WIP: Changes after merge commit 'cb3f9bd' Make the nonlinear beamformer steerable Utilize bitrate above codec max to protect video. Enable VP9 internal resize by default. Filter overlapping RTP header extensions. Make VCMEncodedFrameCallback const. MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback. Remove redudant encoder rate calls. Create isolate files for nonparallel tests. Register header extensions in RtpRtcpObserver to avoid log spam. Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table ACM: Move NACK functionality inside NetEq Fix chromium-style warnings in webrtc/sound/. Create a 'webrtc_nonparallel_tests' target. Update scalability structure data according to updates in the RTP payload profile. audio_coding: rename interface -> include Rewrote perform_action_on_all_files to be parallell. Update reference indices according to updates in the RTP payload profile. Disable P2PTransport...TestFailoverControlledSide on Memcheck pass clangcl compile options to ignore warnings in gflags.cc ...
Diffstat (limited to 'webrtc/video_engine/vie_remb_unittest.cc')
-rw-r--r--webrtc/video_engine/vie_remb_unittest.cc251
1 files changed, 251 insertions, 0 deletions
diff --git a/webrtc/video_engine/vie_remb_unittest.cc b/webrtc/video_engine/vie_remb_unittest.cc
new file mode 100644
index 0000000000..3289c4b822
--- /dev/null
+++ b/webrtc/video_engine/vie_remb_unittest.cc
@@ -0,0 +1,251 @@
+/*
+ * Copyright (c) 2012 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.
+ */
+
+
+// This file includes unit tests for ViERemb.
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include <vector>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
+#include "webrtc/modules/utility/interface/mock/mock_process_thread.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/video_engine/vie_remb.h"
+
+using ::testing::_;
+using ::testing::AnyNumber;
+using ::testing::NiceMock;
+using ::testing::Return;
+
+namespace webrtc {
+
+class ViERembTest : public ::testing::Test {
+ protected:
+ virtual void SetUp() {
+ TickTime::UseFakeClock(12345);
+ process_thread_.reset(new NiceMock<MockProcessThread>);
+ vie_remb_.reset(new VieRemb());
+ }
+ rtc::scoped_ptr<MockProcessThread> process_thread_;
+ rtc::scoped_ptr<VieRemb> vie_remb_;
+};
+
+TEST_F(ViERembTest, OneModuleTestForSendingRemb) {
+ MockRtpRtcp rtp;
+ vie_remb_->AddReceiveChannel(&rtp);
+ vie_remb_->AddRembSender(&rtp);
+
+ const unsigned int bitrate_estimate = 456;
+ unsigned int ssrc = 1234;
+ std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1);
+
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+
+ TickTime::AdvanceFakeClock(1000);
+ EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, ssrcs))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+
+ // Lower bitrate to send another REMB packet.
+ EXPECT_CALL(rtp, SetREMBData(bitrate_estimate - 100, ssrcs))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate - 100);
+
+ vie_remb_->RemoveReceiveChannel(&rtp);
+ vie_remb_->RemoveRembSender(&rtp);
+}
+
+TEST_F(ViERembTest, LowerEstimateToSendRemb) {
+ MockRtpRtcp rtp;
+ vie_remb_->AddReceiveChannel(&rtp);
+ vie_remb_->AddRembSender(&rtp);
+
+ unsigned int bitrate_estimate = 456;
+ unsigned int ssrc = 1234;
+ std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1);
+
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+ // Call OnReceiveBitrateChanged twice to get a first estimate.
+ TickTime::AdvanceFakeClock(1000);
+ EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, ssrcs))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+
+ // Lower the estimate with more than 3% to trigger a call to SetREMBData right
+ // away.
+ bitrate_estimate = bitrate_estimate - 100;
+ EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, ssrcs))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+}
+
+TEST_F(ViERembTest, VerifyIncreasingAndDecreasing) {
+ MockRtpRtcp rtp_0;
+ MockRtpRtcp rtp_1;
+ vie_remb_->AddReceiveChannel(&rtp_0);
+ vie_remb_->AddRembSender(&rtp_0);
+ vie_remb_->AddReceiveChannel(&rtp_1);
+
+ unsigned int bitrate_estimate[] = { 456, 789 };
+ unsigned int ssrc[] = { 1234, 5678 };
+ std::vector<unsigned int> ssrcs(ssrc, ssrc + sizeof(ssrc) / sizeof(ssrc[0]));
+
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate[0]);
+
+ // Call OnReceiveBitrateChanged twice to get a first estimate.
+ EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate[0], ssrcs))
+ .Times(1);
+ TickTime::AdvanceFakeClock(1000);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate[0]);
+
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate[1] + 100);
+
+ // Lower the estimate to trigger a callback.
+ EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate[1], ssrcs))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate[1]);
+
+ vie_remb_->RemoveReceiveChannel(&rtp_0);
+ vie_remb_->RemoveRembSender(&rtp_0);
+ vie_remb_->RemoveReceiveChannel(&rtp_1);
+}
+
+TEST_F(ViERembTest, NoRembForIncreasedBitrate) {
+ MockRtpRtcp rtp_0;
+ MockRtpRtcp rtp_1;
+ vie_remb_->AddReceiveChannel(&rtp_0);
+ vie_remb_->AddRembSender(&rtp_0);
+ vie_remb_->AddReceiveChannel(&rtp_1);
+
+ unsigned int bitrate_estimate = 456;
+ unsigned int ssrc[] = { 1234, 5678 };
+ std::vector<unsigned int> ssrcs(ssrc, ssrc + sizeof(ssrc) / sizeof(ssrc[0]));
+
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+ // Call OnReceiveBitrateChanged twice to get a first estimate.
+ TickTime::AdvanceFakeClock(1000);
+ EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate, ssrcs))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+
+ // Increased estimate shouldn't trigger a callback right away.
+ EXPECT_CALL(rtp_0, SetREMBData(_, _))
+ .Times(0);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate + 1);
+
+ // Decreasing the estimate less than 3% shouldn't trigger a new callback.
+ EXPECT_CALL(rtp_0, SetREMBData(_, _))
+ .Times(0);
+ int lower_estimate = bitrate_estimate * 98 / 100;
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, lower_estimate);
+
+ vie_remb_->RemoveReceiveChannel(&rtp_1);
+ vie_remb_->RemoveReceiveChannel(&rtp_0);
+ vie_remb_->RemoveRembSender(&rtp_0);
+}
+
+TEST_F(ViERembTest, ChangeSendRtpModule) {
+ MockRtpRtcp rtp_0;
+ MockRtpRtcp rtp_1;
+ vie_remb_->AddReceiveChannel(&rtp_0);
+ vie_remb_->AddRembSender(&rtp_0);
+ vie_remb_->AddReceiveChannel(&rtp_1);
+
+ unsigned int bitrate_estimate = 456;
+ unsigned int ssrc[] = { 1234, 5678 };
+ std::vector<unsigned int> ssrcs(ssrc, ssrc + sizeof(ssrc) / sizeof(ssrc[0]));
+
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+ // Call OnReceiveBitrateChanged twice to get a first estimate.
+ TickTime::AdvanceFakeClock(1000);
+ EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate, ssrcs))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+
+ // Decrease estimate to trigger a REMB.
+ bitrate_estimate = bitrate_estimate - 100;
+ EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate, ssrcs))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+
+ // Remove the sending module, add it again -> should get remb on the second
+ // module.
+ vie_remb_->RemoveRembSender(&rtp_0);
+ vie_remb_->AddRembSender(&rtp_1);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+
+ bitrate_estimate = bitrate_estimate - 100;
+ EXPECT_CALL(rtp_1, SetREMBData(bitrate_estimate, ssrcs))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+
+ vie_remb_->RemoveReceiveChannel(&rtp_0);
+ vie_remb_->RemoveReceiveChannel(&rtp_1);
+}
+
+TEST_F(ViERembTest, OnlyOneRembForDoubleProcess) {
+ MockRtpRtcp rtp;
+ unsigned int bitrate_estimate = 456;
+ unsigned int ssrc = 1234;
+ std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1);
+
+ vie_remb_->AddReceiveChannel(&rtp);
+ vie_remb_->AddRembSender(&rtp);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+ // Call OnReceiveBitrateChanged twice to get a first estimate.
+ TickTime::AdvanceFakeClock(1000);
+ EXPECT_CALL(rtp, SetREMBData(_, _))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+
+ // Lower the estimate, should trigger a call to SetREMBData right away.
+ bitrate_estimate = bitrate_estimate - 100;
+ EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, ssrcs))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+
+ // Call OnReceiveBitrateChanged again, this should not trigger a new callback.
+ EXPECT_CALL(rtp, SetREMBData(_, _))
+ .Times(0);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+ vie_remb_->RemoveReceiveChannel(&rtp);
+ vie_remb_->RemoveRembSender(&rtp);
+}
+
+// Only register receiving modules and make sure we fallback to trigger a REMB
+// packet on this one.
+TEST_F(ViERembTest, NoSendingRtpModule) {
+ MockRtpRtcp rtp;
+ vie_remb_->AddReceiveChannel(&rtp);
+
+ unsigned int bitrate_estimate = 456;
+ unsigned int ssrc = 1234;
+ std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1);
+
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+
+ // Call OnReceiveBitrateChanged twice to get a first estimate.
+ TickTime::AdvanceFakeClock(1000);
+ EXPECT_CALL(rtp, SetREMBData(_, _))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+
+ // Lower the estimate to trigger a new packet REMB packet.
+ bitrate_estimate = bitrate_estimate - 100;
+ EXPECT_CALL(rtp, SetREMBData(_, _))
+ .Times(1);
+ vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
+}
+
+} // namespace webrtc