mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00
c8850cbf55
It is meant for Pinpoint to run only the relevant tests when running a bisection. The Pinpoint side of this change can be found here: https://crrev.com/c/2404161 Bug: webrtc:11084 Change-Id: I466f39816b83e2f83a3a49845c99605f4d5a857b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183763 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32082}
98 lines
4 KiB
C++
98 lines
4 KiB
C++
/*
|
|
* 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 "modules/audio_coding/neteq/tools/audio_loop.h"
|
|
#include "rtc_base/format_macros.h"
|
|
#include "rtc_base/time_utils.h"
|
|
#include "test/gtest.h"
|
|
#include "test/testsupport/file_utils.h"
|
|
#include "test/testsupport/perf_test.h"
|
|
|
|
namespace webrtc {
|
|
|
|
namespace {
|
|
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;
|
|
}
|
|
} // namespace
|
|
|
|
// 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);
|
|
|
|
test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_on",
|
|
100.0 * runtime_10999bps / runtime_15500bps, "percent",
|
|
true);
|
|
}
|
|
|
|
// 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);
|
|
|
|
test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_off",
|
|
100.0 * runtime_10999bps / runtime_15500bps, "percent",
|
|
true);
|
|
}
|
|
} // namespace webrtc
|