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a4d873786f
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}
91 lines
2.5 KiB
C++
91 lines
2.5 KiB
C++
/*
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* Copyright 2019 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 "pc/jitter_buffer_delay.h"
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#include <stdint.h>
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#include "absl/types/optional.h"
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#include "api/scoped_refptr.h"
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#include "pc/test/mock_delayable.h"
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#include "rtc_base/ref_counted_object.h"
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#include "rtc_base/thread.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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using ::testing::Return;
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namespace {
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constexpr int kSsrc = 1234;
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} // namespace
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namespace webrtc {
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class JitterBufferDelayTest : public ::testing::Test {
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public:
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JitterBufferDelayTest()
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: delay_(new rtc::RefCountedObject<JitterBufferDelay>(
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rtc::Thread::Current())) {}
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protected:
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rtc::scoped_refptr<JitterBufferDelayInterface> delay_;
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MockDelayable delayable_;
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};
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TEST_F(JitterBufferDelayTest, Set) {
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delay_->OnStart(&delayable_, kSsrc);
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EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 3000))
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.WillOnce(Return(true));
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// Delay in seconds.
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delay_->Set(3.0);
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}
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TEST_F(JitterBufferDelayTest, Caching) {
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// Check that value is cached before start.
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delay_->Set(4.0);
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// Check that cached value applied on the start.
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EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 4000))
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.WillOnce(Return(true));
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delay_->OnStart(&delayable_, kSsrc);
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}
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TEST_F(JitterBufferDelayTest, Clamping) {
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delay_->OnStart(&delayable_, kSsrc);
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// In current Jitter Buffer implementation (Audio or Video) maximum supported
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// value is 10000 milliseconds.
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EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 10000))
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.WillOnce(Return(true));
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delay_->Set(10.5);
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// Test int overflow.
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EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 10000))
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.WillOnce(Return(true));
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delay_->Set(21474836470.0);
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EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 0))
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.WillOnce(Return(true));
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delay_->Set(-21474836470.0);
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// Boundary value in seconds to milliseconds conversion.
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EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 0))
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.WillOnce(Return(true));
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delay_->Set(0.0009);
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EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 0))
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.WillOnce(Return(true));
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delay_->Set(-2.0);
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}
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} // namespace webrtc
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