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7275e18439
We've done this previously with the other audio encoders, but Opus had to wait until all external users had been updated. BUG=webrtc:7847 Change-Id: I70422d7b6c715f32a43bee88febcf6b6155e18b3 Reviewed-on: https://webrtc-review.googlesource.com/8000 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20424}
98 lines
4.1 KiB
C++
98 lines
4.1 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "api/audio_codecs/opus/audio_encoder_opus.h"
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#include "modules/audio_coding/neteq/tools/audio_loop.h"
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#include "rtc_base/format_macros.h"
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#include "rtc_base/timeutils.h"
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#include "test/gtest.h"
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#include "test/testsupport/fileutils.h"
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#include "test/testsupport/perf_test.h"
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namespace webrtc {
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namespace {
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int64_t RunComplexityTest(const AudioEncoderOpusConfig& config) {
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// Create encoder.
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constexpr int payload_type = 17;
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const auto encoder = AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
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// Open speech file.
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const std::string kInputFileName =
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webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm");
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test::AudioLoop audio_loop;
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constexpr int kSampleRateHz = 48000;
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EXPECT_EQ(kSampleRateHz, encoder->SampleRateHz());
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constexpr size_t kMaxLoopLengthSamples =
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kSampleRateHz * 10; // 10 second loop.
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constexpr size_t kInputBlockSizeSamples =
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10 * kSampleRateHz / 1000; // 60 ms.
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EXPECT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
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kInputBlockSizeSamples));
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// Encode.
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const int64_t start_time_ms = rtc::TimeMillis();
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AudioEncoder::EncodedInfo info;
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rtc::Buffer encoded(500);
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uint32_t rtp_timestamp = 0u;
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for (size_t i = 0; i < 10000; ++i) {
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encoded.Clear();
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info = encoder->Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded);
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rtp_timestamp += kInputBlockSizeSamples;
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}
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return rtc::TimeMillis() - start_time_ms;
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}
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} // namespace
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// This test encodes an audio file using Opus twice with different bitrates
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// (~11 kbps and 15.5 kbps). The runtime for each is measured, and the ratio
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// between the two is calculated and tracked. This test explicitly sets the
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// low_rate_complexity to 9. When running on desktop platforms, this is the same
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// as the regular complexity, and the expectation is that the resulting ratio
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// should be less than 100% (since the encoder runs faster at lower bitrates,
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// given a fixed complexity setting). On the other hand, when running on
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// mobiles, the regular complexity is 5, and we expect the resulting ratio to
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// be higher, since we have explicitly asked for a higher complexity setting at
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// the lower rate.
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TEST(AudioEncoderOpusComplexityAdaptationTest, AdaptationOn) {
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// Create config.
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AudioEncoderOpusConfig config;
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// The limit -- including the hysteresis window -- at which the complexity
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// shuold be increased.
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config.bitrate_bps = rtc::Optional<int>(11000 - 1);
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config.low_rate_complexity = 9;
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int64_t runtime_10999bps = RunComplexityTest(config);
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config.bitrate_bps = rtc::Optional<int>(15500);
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int64_t runtime_15500bps = RunComplexityTest(config);
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test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_on",
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100.0 * runtime_10999bps / runtime_15500bps, "percent",
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true);
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}
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// This test is identical to the one above, but without the complexity
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// adaptation enabled (neither on desktop, nor on mobile). The expectation is
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// that the resulting ratio is less than 100% at all times.
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TEST(AudioEncoderOpusComplexityAdaptationTest, AdaptationOff) {
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// Create config.
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AudioEncoderOpusConfig config;
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// The limit -- including the hysteresis window -- at which the complexity
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// shuold be increased (but not in this test since complexity adaptation is
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// disabled).
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config.bitrate_bps = rtc::Optional<int>(11000 - 1);
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int64_t runtime_10999bps = RunComplexityTest(config);
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config.bitrate_bps = rtc::Optional<int>(15500);
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int64_t runtime_15500bps = RunComplexityTest(config);
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test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_off",
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100.0 * runtime_10999bps / runtime_15500bps, "percent",
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true);
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}
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} // namespace webrtc
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