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Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc | 105 |
1 files changed, 105 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc new file mode 100644 index 0000000000..e8c131092c --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc @@ -0,0 +1,105 @@ +/* + * 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 |