On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface, GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes the same concept. Bug: webrtc:10287 Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957 Reviewed-on: https://webrtc-review.googlesource.com/c/123482 Commit-Queue: Ruslan Burakov <kuddai@google.com> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26877}
104 lines
3.0 KiB
C++
104 lines
3.0 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 <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/playout_latency.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 PlayoutLatencyTest : public testing::Test {
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public:
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PlayoutLatencyTest()
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: latency_(
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new rtc::RefCountedObject<PlayoutLatency>(rtc::Thread::Current())) {
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}
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protected:
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rtc::scoped_refptr<PlayoutLatencyInterface> latency_;
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MockDelayable delayable_;
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};
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TEST_F(PlayoutLatencyTest, DefaultValue) {
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EXPECT_DOUBLE_EQ(0.0, latency_->GetLatency());
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}
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TEST_F(PlayoutLatencyTest, GetLatency) {
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latency_->OnStart(&delayable_, kSsrc);
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EXPECT_CALL(delayable_, GetBaseMinimumPlayoutDelayMs(kSsrc))
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.WillOnce(Return(2000));
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// Latency in seconds.
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EXPECT_DOUBLE_EQ(2.0, latency_->GetLatency());
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EXPECT_CALL(delayable_, GetBaseMinimumPlayoutDelayMs(kSsrc))
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.WillOnce(Return(absl::nullopt));
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// When no value is returned by GetBaseMinimumPlayoutDelayMs, and there are
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// no caching, then return default value.
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EXPECT_DOUBLE_EQ(0.0, latency_->GetLatency());
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}
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TEST_F(PlayoutLatencyTest, SetLatency) {
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latency_->OnStart(&delayable_, kSsrc);
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EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 3000))
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.WillOnce(Return(true));
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// Latency in seconds.
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latency_->SetLatency(3.0);
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}
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TEST_F(PlayoutLatencyTest, Caching) {
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// Check that value is cached before start.
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latency_->SetLatency(4.0);
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// Latency in seconds.
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EXPECT_DOUBLE_EQ(4.0, latency_->GetLatency());
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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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latency_->OnStart(&delayable_, kSsrc);
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EXPECT_CALL(delayable_, GetBaseMinimumPlayoutDelayMs(kSsrc))
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.WillOnce(Return(absl::nullopt));
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// On false the latest cached value is returned.
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EXPECT_DOUBLE_EQ(4.0, latency_->GetLatency());
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latency_->OnStop();
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// Check that after stop it returns last cached value.
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EXPECT_DOUBLE_EQ(4.0, latency_->GetLatency());
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}
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TEST_F(PlayoutLatencyTest, Rounding) {
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latency_->OnStart(&delayable_, kSsrc);
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// In Jitter Buffer (Audio or Video) delay 0 has a special meaning of
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// unconstrained variable, that is why here if latency is small enough we
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// round it to 0 delay.
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EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 0))
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.WillOnce(Return(true));
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latency_->SetLatency(0.005);
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}
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} // namespace webrtc
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