webrtc_m130/pc/playout_latency_unittest.cc
Ruslan Burakov 493a650b1e Propagate base minimum delay from video jitter buffer to webrtc/api.
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}
2019-02-27 15:08:34 +00:00

104 lines
3.0 KiB
C++

/*
* Copyright 2019 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 <stdint.h>
#include "absl/types/optional.h"
#include "api/scoped_refptr.h"
#include "pc/playout_latency.h"
#include "pc/test/mock_delayable.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/thread.h"
#include "test/gmock.h"
#include "test/gtest.h"
using ::testing::Return;
namespace {
constexpr int kSsrc = 1234;
} // namespace
namespace webrtc {
class PlayoutLatencyTest : public testing::Test {
public:
PlayoutLatencyTest()
: latency_(
new rtc::RefCountedObject<PlayoutLatency>(rtc::Thread::Current())) {
}
protected:
rtc::scoped_refptr<PlayoutLatencyInterface> latency_;
MockDelayable delayable_;
};
TEST_F(PlayoutLatencyTest, DefaultValue) {
EXPECT_DOUBLE_EQ(0.0, latency_->GetLatency());
}
TEST_F(PlayoutLatencyTest, GetLatency) {
latency_->OnStart(&delayable_, kSsrc);
EXPECT_CALL(delayable_, GetBaseMinimumPlayoutDelayMs(kSsrc))
.WillOnce(Return(2000));
// Latency in seconds.
EXPECT_DOUBLE_EQ(2.0, latency_->GetLatency());
EXPECT_CALL(delayable_, GetBaseMinimumPlayoutDelayMs(kSsrc))
.WillOnce(Return(absl::nullopt));
// When no value is returned by GetBaseMinimumPlayoutDelayMs, and there are
// no caching, then return default value.
EXPECT_DOUBLE_EQ(0.0, latency_->GetLatency());
}
TEST_F(PlayoutLatencyTest, SetLatency) {
latency_->OnStart(&delayable_, kSsrc);
EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 3000))
.WillOnce(Return(true));
// Latency in seconds.
latency_->SetLatency(3.0);
}
TEST_F(PlayoutLatencyTest, Caching) {
// Check that value is cached before start.
latency_->SetLatency(4.0);
// Latency in seconds.
EXPECT_DOUBLE_EQ(4.0, latency_->GetLatency());
// Check that cached value applied on the start.
EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 4000))
.WillOnce(Return(true));
latency_->OnStart(&delayable_, kSsrc);
EXPECT_CALL(delayable_, GetBaseMinimumPlayoutDelayMs(kSsrc))
.WillOnce(Return(absl::nullopt));
// On false the latest cached value is returned.
EXPECT_DOUBLE_EQ(4.0, latency_->GetLatency());
latency_->OnStop();
// Check that after stop it returns last cached value.
EXPECT_DOUBLE_EQ(4.0, latency_->GetLatency());
}
TEST_F(PlayoutLatencyTest, Rounding) {
latency_->OnStart(&delayable_, kSsrc);
// In Jitter Buffer (Audio or Video) delay 0 has a special meaning of
// unconstrained variable, that is why here if latency is small enough we
// round it to 0 delay.
EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 0))
.WillOnce(Return(true));
latency_->SetLatency(0.005);
}
} // namespace webrtc