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/*
* 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/main/source/media_optimization.h"
#include "webrtc/system_wrappers/interface/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 = kKeyFrame;
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,
NULL,
NULL); // 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,
NULL,
NULL); // 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,
NULL,
NULL); // 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,
NULL,
NULL); // 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
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