blob: 27e58ad24024deac005288affa247d7ce459b167 [file] [log] [blame]
/*
* Copyright (c) 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 "video/encoder_overshoot_detector.h"
#include <string>
#include "api/units/data_rate.h"
#include "rtc_base/fake_clock.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/metrics.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
using ::testing::TestWithParam;
using ::testing::ValuesIn;
static std::string CodecTypeToHistogramSuffix(VideoCodecType codec) {
switch (codec) {
case kVideoCodecVP8:
return "Vp8";
case kVideoCodecVP9:
return "Vp9";
case kVideoCodecAV1:
return "Av1";
case kVideoCodecH264:
return "H264";
case kVideoCodecH265:
return "H265";
case kVideoCodecGeneric:
return "Generic";
}
}
struct TestParams {
VideoCodecType codec_type;
bool is_screenshare;
};
} // namespace
class EncoderOvershootDetectorTest : public TestWithParam<TestParams> {
public:
static constexpr int kDefaultBitrateBps = 300000;
static constexpr double kDefaultFrameRateFps = 15;
EncoderOvershootDetectorTest()
: detector_(kWindowSizeMs,
GetParam().codec_type,
GetParam().is_screenshare),
target_bitrate_(DataRate::BitsPerSec(kDefaultBitrateBps)),
target_framerate_fps_(kDefaultFrameRateFps) {}
protected:
void SetUp() override { metrics::Reset(); }
void RunConstantUtilizationTest(double actual_utilization_factor,
double expected_utilization_factor,
double allowed_error,
int64_t test_duration_ms) {
const int frame_size_bytes =
static_cast<int>(actual_utilization_factor *
(target_bitrate_.bps() / target_framerate_fps_) / 8);
detector_.SetTargetRate(target_bitrate_, target_framerate_fps_,
rtc::TimeMillis());
if (rtc::TimeMillis() == 0) {
// Encode a first frame which by definition has no overuse factor.
detector_.OnEncodedFrame(frame_size_bytes, rtc::TimeMillis());
clock_.AdvanceTime(TimeDelta::Seconds(1) / target_framerate_fps_);
}
int64_t runtime_us = 0;
while (runtime_us < test_duration_ms * 1000) {
detector_.OnEncodedFrame(frame_size_bytes, rtc::TimeMillis());
runtime_us += rtc::kNumMicrosecsPerSec / target_framerate_fps_;
clock_.AdvanceTime(TimeDelta::Seconds(1) / target_framerate_fps_);
}
// At constant utilization, both network and media utilization should be
// close to expected.
const absl::optional<double> network_utilization_factor =
detector_.GetNetworkRateUtilizationFactor(rtc::TimeMillis());
EXPECT_NEAR(network_utilization_factor.value_or(-1),
expected_utilization_factor, allowed_error);
const absl::optional<double> media_utilization_factor =
detector_.GetMediaRateUtilizationFactor(rtc::TimeMillis());
EXPECT_NEAR(media_utilization_factor.value_or(-1),
expected_utilization_factor, allowed_error);
}
static constexpr int64_t kWindowSizeMs = 3000;
EncoderOvershootDetector detector_;
rtc::ScopedFakeClock clock_;
DataRate target_bitrate_;
double target_framerate_fps_;
};
TEST_P(EncoderOvershootDetectorTest, NoUtilizationIfNoRate) {
const int frame_size_bytes = 1000;
const int64_t time_interval_ms = 33;
detector_.SetTargetRate(target_bitrate_, target_framerate_fps_,
rtc::TimeMillis());
// No data points, can't determine overshoot rate.
EXPECT_FALSE(
detector_.GetNetworkRateUtilizationFactor(rtc::TimeMillis()).has_value());
detector_.OnEncodedFrame(frame_size_bytes, rtc::TimeMillis());
clock_.AdvanceTime(TimeDelta::Millis(time_interval_ms));
EXPECT_TRUE(
detector_.GetNetworkRateUtilizationFactor(rtc::TimeMillis()).has_value());
}
TEST_P(EncoderOvershootDetectorTest, OptimalSize) {
// Optimally behaved encoder.
// Allow some error margin due to rounding errors, eg due to frame
// interval not being an integer.
RunConstantUtilizationTest(1.0, 1.0, 0.01, kWindowSizeMs);
}
TEST_P(EncoderOvershootDetectorTest, Undershoot) {
// Undershoot, reported utilization factor should be capped to 1.0 so
// that we don't incorrectly boost encoder bitrate during movement.
RunConstantUtilizationTest(0.5, 1.0, 0.00, kWindowSizeMs);
}
TEST_P(EncoderOvershootDetectorTest, Overshoot) {
// Overshoot by 20%.
// Allow some error margin due to rounding errors.
RunConstantUtilizationTest(1.2, 1.2, 0.01, kWindowSizeMs);
}
TEST_P(EncoderOvershootDetectorTest, ConstantOvershootVaryingRates) {
// Overshoot by 20%, but vary framerate and bitrate.
// Allow some error margin due to rounding errors.
RunConstantUtilizationTest(1.2, 1.2, 0.01, kWindowSizeMs);
target_framerate_fps_ /= 2;
RunConstantUtilizationTest(1.2, 1.2, 0.01, kWindowSizeMs / 2);
target_bitrate_ = DataRate::BitsPerSec(target_bitrate_.bps() / 2);
RunConstantUtilizationTest(1.2, 1.2, 0.01, kWindowSizeMs / 2);
}
TEST_P(EncoderOvershootDetectorTest, ConstantRateVaryingOvershoot) {
// Overshoot by 10%, keep framerate and bitrate constant.
// Allow some error margin due to rounding errors.
RunConstantUtilizationTest(1.1, 1.1, 0.01, kWindowSizeMs);
// Change overshoot to 20%, run for half window and expect overshoot
// to be 15%.
RunConstantUtilizationTest(1.2, 1.15, 0.01, kWindowSizeMs / 2);
// Keep running at 20% overshoot, after window is full that should now
// be the reported overshoot.
RunConstantUtilizationTest(1.2, 1.2, 0.01, kWindowSizeMs / 2);
}
TEST_P(EncoderOvershootDetectorTest, PartialOvershoot) {
const int ideal_frame_size_bytes =
(target_bitrate_.bps() / target_framerate_fps_) / 8;
detector_.SetTargetRate(target_bitrate_, target_framerate_fps_,
rtc::TimeMillis());
// Test scenario with average bitrate matching the target bitrate, but
// with some utilization factor penalty as the frames can't be paced out
// on the network at the target rate.
// Insert a series of four frames:
// 1) 20% overshoot, not penalized as buffer if empty.
// 2) 20% overshoot, the 20% overshoot from the first frame is penalized.
// 3) 20% undershoot, negating the overshoot from the last frame.
// 4) 20% undershoot, no penalty.
// On average then utilization penalty is thus 5%.
int64_t runtime_us = 0;
int i = 0;
while (runtime_us < kWindowSizeMs * rtc::kNumMicrosecsPerMillisec) {
runtime_us += rtc::kNumMicrosecsPerSec / target_framerate_fps_;
clock_.AdvanceTime(TimeDelta::Seconds(1) / target_framerate_fps_);
int frame_size_bytes = (i++ % 4 < 2) ? (ideal_frame_size_bytes * 120) / 100
: (ideal_frame_size_bytes * 80) / 100;
detector_.OnEncodedFrame(frame_size_bytes, rtc::TimeMillis());
}
// Expect 5% overshoot for network rate, see above.
const absl::optional<double> network_utilization_factor =
detector_.GetNetworkRateUtilizationFactor(rtc::TimeMillis());
EXPECT_NEAR(network_utilization_factor.value_or(-1), 1.05, 0.01);
// Expect media rate to be on average correct.
const absl::optional<double> media_utilization_factor =
detector_.GetMediaRateUtilizationFactor(rtc::TimeMillis());
EXPECT_NEAR(media_utilization_factor.value_or(-1), 1.00, 0.01);
}
TEST_P(EncoderOvershootDetectorTest, RecordsZeroErrorMetricWithNoOvershoot) {
DataSize ideal_frame_size =
target_bitrate_ / Frequency::Hertz(target_framerate_fps_);
detector_.SetTargetRate(target_bitrate_, target_framerate_fps_,
rtc::TimeMillis());
detector_.OnEncodedFrame(ideal_frame_size.bytes(), rtc::TimeMillis());
detector_.Reset();
const VideoCodecType codec = GetParam().codec_type;
const bool is_screenshare = GetParam().is_screenshare;
const std::string rmse_histogram_prefix =
is_screenshare ? "WebRTC.Video.Screenshare.RMSEOfEncodingBitrateInKbps."
: "WebRTC.Video.RMSEOfEncodingBitrateInKbps.";
const std::string overshoot_histogram_prefix =
is_screenshare ? "WebRTC.Video.Screenshare.EncodingBitrateOvershoot."
: "WebRTC.Video.EncodingBitrateOvershoot.";
// RMSE and overshoot percent = 0, since we used ideal frame size.
EXPECT_METRIC_EQ(1, metrics::NumSamples(rmse_histogram_prefix +
CodecTypeToHistogramSuffix(codec)));
EXPECT_METRIC_EQ(
1, metrics::NumEvents(
rmse_histogram_prefix + CodecTypeToHistogramSuffix(codec), 0));
EXPECT_METRIC_EQ(1, metrics::NumSamples(overshoot_histogram_prefix +
CodecTypeToHistogramSuffix(codec)));
EXPECT_METRIC_EQ(1, metrics::NumEvents(overshoot_histogram_prefix +
CodecTypeToHistogramSuffix(codec),
0));
}
TEST_P(EncoderOvershootDetectorTest,
RecordScreenshareZeroMetricWithNoOvershoot) {
DataSize ideal_frame_size =
target_bitrate_ / Frequency::Hertz(target_framerate_fps_);
// Use target frame size with 50% overshoot.
DataSize target_frame_size = ideal_frame_size * 3 / 2;
detector_.SetTargetRate(target_bitrate_, target_framerate_fps_,
rtc::TimeMillis());
detector_.OnEncodedFrame(target_frame_size.bytes(), rtc::TimeMillis());
detector_.Reset();
const VideoCodecType codec = GetParam().codec_type;
const bool is_screenshare = GetParam().is_screenshare;
const std::string rmse_histogram_prefix =
is_screenshare ? "WebRTC.Video.Screenshare.RMSEOfEncodingBitrateInKbps."
: "WebRTC.Video.RMSEOfEncodingBitrateInKbps.";
const std::string overshoot_histogram_prefix =
is_screenshare ? "WebRTC.Video.Screenshare.EncodingBitrateOvershoot."
: "WebRTC.Video.EncodingBitrateOvershoot.";
// Use ideal_frame_size_kbits to represnt ideal_frame_size.bytes()*8/1000,
// then rmse_in_kbps = ideal_frame_size_kbits/2
// since we use target frame size with 50% overshoot.
int64_t rmse_in_kbps = ideal_frame_size.bytes() * 8 / 1000 / 2;
EXPECT_METRIC_EQ(1, metrics::NumSamples(rmse_histogram_prefix +
CodecTypeToHistogramSuffix(codec)));
EXPECT_METRIC_EQ(1, metrics::NumEvents(rmse_histogram_prefix +
CodecTypeToHistogramSuffix(codec),
rmse_in_kbps));
// overshoot percent = 50, since we used ideal_frame_size * 3 / 2;
EXPECT_METRIC_EQ(1, metrics::NumSamples(overshoot_histogram_prefix +
CodecTypeToHistogramSuffix(codec)));
EXPECT_METRIC_EQ(1, metrics::NumEvents(overshoot_histogram_prefix +
CodecTypeToHistogramSuffix(codec),
50));
}
INSTANTIATE_TEST_SUITE_P(
PerCodecType,
EncoderOvershootDetectorTest,
ValuesIn<TestParams>({{VideoCodecType::kVideoCodecVP8, false},
{VideoCodecType::kVideoCodecVP8, true},
{VideoCodecType::kVideoCodecVP9, false},
{VideoCodecType::kVideoCodecVP9, true},
{VideoCodecType::kVideoCodecAV1, false},
{VideoCodecType::kVideoCodecAV1, true},
{VideoCodecType::kVideoCodecH264, false},
{VideoCodecType::kVideoCodecH264, true},
{VideoCodecType::kVideoCodecH265, false},
{VideoCodecType::kVideoCodecH265, true}}));
} // namespace webrtc