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/*
* Copyright (c) 2016 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 "api/audio_codecs/opus/audio_encoder_opus.h"
#include "api/test/metrics/global_metrics_logger_and_exporter.h"
#include "api/test/metrics/metric.h"
#include "modules/audio_coding/neteq/tools/audio_loop.h"
#include "rtc_base/time_utils.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
namespace {
using ::webrtc::test::GetGlobalMetricsLogger;
using ::webrtc::test::ImprovementDirection;
using ::webrtc::test::Unit;
int64_t RunComplexityTest(const AudioEncoderOpusConfig& config) {
// Create encoder.
constexpr int payload_type = 17;
const auto encoder = AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
// Open speech file.
const std::string kInputFileName =
webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm");
test::AudioLoop audio_loop;
constexpr int kSampleRateHz = 48000;
EXPECT_EQ(kSampleRateHz, encoder->SampleRateHz());
constexpr size_t kMaxLoopLengthSamples =
kSampleRateHz * 10; // 10 second loop.
constexpr size_t kInputBlockSizeSamples =
10 * kSampleRateHz / 1000; // 60 ms.
EXPECT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
kInputBlockSizeSamples));
// Encode.
const int64_t start_time_ms = rtc::TimeMillis();
AudioEncoder::EncodedInfo info;
rtc::Buffer encoded(500);
uint32_t rtp_timestamp = 0u;
for (size_t i = 0; i < 10000; ++i) {
encoded.Clear();
info = encoder->Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded);
rtp_timestamp += kInputBlockSizeSamples;
}
return rtc::TimeMillis() - start_time_ms;
}
// This test encodes an audio file using Opus twice with different bitrates
// (~11 kbps and 15.5 kbps). The runtime for each is measured, and the ratio
// between the two is calculated and tracked. This test explicitly sets the
// low_rate_complexity to 9. When running on desktop platforms, this is the same
// as the regular complexity, and the expectation is that the resulting ratio
// should be less than 100% (since the encoder runs faster at lower bitrates,
// given a fixed complexity setting). On the other hand, when running on
// mobiles, the regular complexity is 5, and we expect the resulting ratio to
// be higher, since we have explicitly asked for a higher complexity setting at
// the lower rate.
TEST(AudioEncoderOpusComplexityAdaptationTest, Adaptation_On) {
// Create config.
AudioEncoderOpusConfig config;
// The limit -- including the hysteresis window -- at which the complexity
// shuold be increased.
config.bitrate_bps = 11000 - 1;
config.low_rate_complexity = 9;
int64_t runtime_10999bps = RunComplexityTest(config);
config.bitrate_bps = 15500;
int64_t runtime_15500bps = RunComplexityTest(config);
GetGlobalMetricsLogger()->LogSingleValueMetric(
"opus_encoding_complexity_ratio", "adaptation_on",
100.0 * runtime_10999bps / runtime_15500bps, Unit::kPercent,
ImprovementDirection::kNeitherIsBetter);
}
// This test is identical to the one above, but without the complexity
// adaptation enabled (neither on desktop, nor on mobile). The expectation is
// that the resulting ratio is less than 100% at all times.
TEST(AudioEncoderOpusComplexityAdaptationTest, Adaptation_Off) {
// Create config.
AudioEncoderOpusConfig config;
// The limit -- including the hysteresis window -- at which the complexity
// shuold be increased (but not in this test since complexity adaptation is
// disabled).
config.bitrate_bps = 11000 - 1;
int64_t runtime_10999bps = RunComplexityTest(config);
config.bitrate_bps = 15500;
int64_t runtime_15500bps = RunComplexityTest(config);
GetGlobalMetricsLogger()->LogSingleValueMetric(
"opus_encoding_complexity_ratio", "adaptation_off",
100.0 * runtime_10999bps / runtime_15500bps, Unit::kPercent,
ImprovementDirection::kNeitherIsBetter);
}
} // namespace
} // namespace webrtc
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