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Merge two files with AudioEncoderOpus tests

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Merge the contents of audio_encoder_mutable_opus_test.cc into
audio_encoder_opus_unittest.cc, since they're now both testing
AudioEncoderOpus.

(While preparing this CL, I noted a bug in the PacketLossRateOptimized
test. This CL leaves that test essentially unchanged; I've posted bug
4981 about the problem.)

Review URL: https://codereview.webrtc.org/1319713004

Cr-Commit-Position: refs/heads/master@{#9906}
This commit is contained in:
kwiberg
2015-09-09 04:38:32 -07:00
committed by Commit bot
parent 529528cc36
commit 77d22fa014
3 changed files with 114 additions and 143 deletions
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109
webrtc/modules/audio_coding/codecs/opus/audio_encoder_mutable_opus_test.cc
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@ -1,109 +0,0 @@
/*
* Copyright (c) 2015 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.
*/
// TODO(kwiberg): Merge these tests into audio_encoder_opus_unittest.cc
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
namespace webrtc {
namespace acm2 {
#ifdef WEBRTC_CODEC_OPUS
namespace {
const CodecInst kDefaultOpusCodecInst = {105, "opus", 48000, 960, 1, 32000};
} // namespace
class AudioEncoderMutableOpusTest : public ::testing::Test {
protected:
AudioEncoderMutableOpusTest() : codec_inst_(kDefaultOpusCodecInst) {}
void CreateCodec(int num_channels) {
codec_inst_.channels = num_channels;
encoder_.reset(new AudioEncoderOpus(codec_inst_));
auto expected_app =
num_channels == 1 ? AudioEncoderOpus::kVoip : AudioEncoderOpus::kAudio;
EXPECT_EQ(expected_app, encoder_->application());
}
CodecInst codec_inst_;
rtc::scoped_ptr<AudioEncoderOpus> encoder_;
};
TEST_F(AudioEncoderMutableOpusTest, DefaultApplicationModeMono) {
CreateCodec(1);
}
TEST_F(AudioEncoderMutableOpusTest, DefaultApplicationModeStereo) {
CreateCodec(2);
}
TEST_F(AudioEncoderMutableOpusTest, ChangeApplicationMode) {
CreateCodec(2);
EXPECT_TRUE(encoder_->SetApplication(AudioEncoder::Application::kSpeech));
EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
}
TEST_F(AudioEncoderMutableOpusTest, ResetWontChangeApplicationMode) {
CreateCodec(2);
// Trigger a reset.
encoder_->Reset();
// Verify that the mode is still kAudio.
EXPECT_EQ(AudioEncoderOpus::kAudio, encoder_->application());
// Now change to kVoip.
EXPECT_TRUE(encoder_->SetApplication(AudioEncoder::Application::kSpeech));
EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
// Trigger a reset again.
encoder_->Reset();
// Verify that the mode is still kVoip.
EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
}
TEST_F(AudioEncoderMutableOpusTest, ToggleDtx) {
CreateCodec(2);
// Enable DTX
EXPECT_TRUE(encoder_->SetDtx(true));
// Verify that the mode is still kAudio.
EXPECT_EQ(AudioEncoderOpus::kAudio, encoder_->application());
// Turn off DTX.
EXPECT_TRUE(encoder_->SetDtx(false));
}
TEST_F(AudioEncoderMutableOpusTest, SetBitrate) {
CreateCodec(1);
// Constants are replicated from audio_encoder_opus.cc.
const int kMinBitrateBps = 500;
const int kMaxBitrateBps = 512000;
// Set a too low bitrate.
encoder_->SetTargetBitrate(kMinBitrateBps - 1);
EXPECT_EQ(kMinBitrateBps, encoder_->GetTargetBitrate());
// Set a too high bitrate.
encoder_->SetTargetBitrate(kMaxBitrateBps + 1);
EXPECT_EQ(kMaxBitrateBps, encoder_->GetTargetBitrate());
// Set the minimum rate.
encoder_->SetTargetBitrate(kMinBitrateBps);
EXPECT_EQ(kMinBitrateBps, encoder_->GetTargetBitrate());
// Set the maximum rate.
encoder_->SetTargetBitrate(kMaxBitrateBps);
EXPECT_EQ(kMaxBitrateBps, encoder_->GetTargetBitrate());
// Set rates from 1000 up to 32000 bps.
for (int rate = 1000; rate <= 32000; rate += 1000) {
encoder_->SetTargetBitrate(rate);
EXPECT_EQ(rate, encoder_->GetTargetBitrate());
}
}
#endif // WEBRTC_CODEC_OPUS
} // namespace acm2
} // namespace webrtc

147
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
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@ -10,32 +10,97 @@
#include "testing/gtest/include/gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h" #include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h" #include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
namespace webrtc { namespace webrtc {
namespace {
const CodecInst kOpusSettings = {105, "opus", 48000, 960, 1, 32000};
} // namespace
class AudioEncoderOpusTest : public ::testing::Test { class AudioEncoderOpusTest : public ::testing::Test {
protected: protected:
// The constructor simply creates an Opus encoder with default configuration. void CreateCodec(int num_channels) {
AudioEncoderOpusTest() codec_inst_.channels = num_channels;
: opus_(new AudioEncoderOpus(AudioEncoderOpus::Config())) {} encoder_.reset(new AudioEncoderOpus(codec_inst_));
auto expected_app =
// Repeatedly sets packet loss rates in the range [from, to], increasing by num_channels == 1 ? AudioEncoderOpus::kVoip : AudioEncoderOpus::kAudio;
// 0.01 in each step. The function verifies that the actual loss rate is EXPECT_EQ(expected_app, encoder_->application());
// |expected_return|.
void TestSetPacketLossRate(double from, double to, double expected_return) {
ASSERT_TRUE(opus_);
for (double loss = from; loss <= to;
(to >= from) ? loss += 0.01 : loss -= 0.01) {
opus_->SetProjectedPacketLossRate(loss);
EXPECT_DOUBLE_EQ(expected_return, opus_->packet_loss_rate());
}
} }
rtc::scoped_ptr<AudioEncoderOpus> opus_; CodecInst codec_inst_ = kOpusSettings;
rtc::scoped_ptr<AudioEncoderOpus> encoder_;
}; };
TEST_F(AudioEncoderOpusTest, DefaultApplicationModeMono) {
CreateCodec(1);
}
TEST_F(AudioEncoderOpusTest, DefaultApplicationModeStereo) {
CreateCodec(2);
}
TEST_F(AudioEncoderOpusTest, ChangeApplicationMode) {
CreateCodec(2);
EXPECT_TRUE(encoder_->SetApplication(AudioEncoder::Application::kSpeech));
EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
}
TEST_F(AudioEncoderOpusTest, ResetWontChangeApplicationMode) {
CreateCodec(2);
// Trigger a reset.
encoder_->Reset();
// Verify that the mode is still kAudio.
EXPECT_EQ(AudioEncoderOpus::kAudio, encoder_->application());
// Now change to kVoip.
EXPECT_TRUE(encoder_->SetApplication(AudioEncoder::Application::kSpeech));
EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
// Trigger a reset again.
encoder_->Reset();
// Verify that the mode is still kVoip.
EXPECT_EQ(AudioEncoderOpus::kVoip, encoder_->application());
}
TEST_F(AudioEncoderOpusTest, ToggleDtx) {
CreateCodec(2);
// Enable DTX
EXPECT_TRUE(encoder_->SetDtx(true));
// Verify that the mode is still kAudio.
EXPECT_EQ(AudioEncoderOpus::kAudio, encoder_->application());
// Turn off DTX.
EXPECT_TRUE(encoder_->SetDtx(false));
}
TEST_F(AudioEncoderOpusTest, SetBitrate) {
CreateCodec(1);
// Constants are replicated from audio_encoder_opus.cc.
const int kMinBitrateBps = 500;
const int kMaxBitrateBps = 512000;
// Set a too low bitrate.
encoder_->SetTargetBitrate(kMinBitrateBps - 1);
EXPECT_EQ(kMinBitrateBps, encoder_->GetTargetBitrate());
// Set a too high bitrate.
encoder_->SetTargetBitrate(kMaxBitrateBps + 1);
EXPECT_EQ(kMaxBitrateBps, encoder_->GetTargetBitrate());
// Set the minimum rate.
encoder_->SetTargetBitrate(kMinBitrateBps);
EXPECT_EQ(kMinBitrateBps, encoder_->GetTargetBitrate());
// Set the maximum rate.
encoder_->SetTargetBitrate(kMaxBitrateBps);
EXPECT_EQ(kMaxBitrateBps, encoder_->GetTargetBitrate());
// Set rates from 1000 up to 32000 bps.
for (int rate = 1000; rate <= 32000; rate += 1000) {
encoder_->SetTargetBitrate(rate);
EXPECT_EQ(rate, encoder_->GetTargetBitrate());
}
}
namespace { namespace {
// These constants correspond to those used in // These constants correspond to those used in
// AudioEncoderOpus::SetProjectedPacketLossRate. // AudioEncoderOpus::SetProjectedPacketLossRate.
const double kPacketLossRate20 = 0.20; const double kPacketLossRate20 = 0.20;
@ -45,36 +110,52 @@ const double kPacketLossRate1 = 0.01;
const double kLossRate20Margin = 0.02; const double kLossRate20Margin = 0.02;
const double kLossRate10Margin = 0.01; const double kLossRate10Margin = 0.01;
const double kLossRate5Margin = 0.01; const double kLossRate5Margin = 0.01;
// Repeatedly sets packet loss rates in the range [from, to], increasing by
// 0.01 in each step. The function verifies that the actual loss rate is
// |expected_return|.
void TestSetPacketLossRate(AudioEncoderOpus* encoder,
double from,
double to,
double expected_return) {
for (double loss = from; loss <= to;
(to >= from) ? loss += 0.01 : loss -= 0.01) {
encoder->SetProjectedPacketLossRate(loss);
EXPECT_DOUBLE_EQ(expected_return, encoder->packet_loss_rate());
}
}
} // namespace } // namespace
TEST_F(AudioEncoderOpusTest, PacketLossRateOptimized) { TEST_F(AudioEncoderOpusTest, PacketLossRateOptimized) {
CreateCodec(1);
// Note that the order of the following calls is critical. // Note that the order of the following calls is critical.
TestSetPacketLossRate(0.0, 0.0, 0.0); TestSetPacketLossRate(encoder_.get(), 0.0, 0.0, 0.0);
TestSetPacketLossRate(kPacketLossRate1, TestSetPacketLossRate(encoder_.get(), kPacketLossRate1,
kPacketLossRate5 + kLossRate5Margin - 0.01, kPacketLossRate5 + kLossRate5Margin - 0.01,
kPacketLossRate1); kPacketLossRate1);
TestSetPacketLossRate(kPacketLossRate5 + kLossRate5Margin, TestSetPacketLossRate(encoder_.get(), kPacketLossRate5 + kLossRate5Margin,
kPacketLossRate10 + kLossRate10Margin - 0.01, kPacketLossRate10 + kLossRate10Margin - 0.01,
kPacketLossRate5); kPacketLossRate5);
TestSetPacketLossRate(kPacketLossRate10 + kLossRate10Margin, TestSetPacketLossRate(encoder_.get(), kPacketLossRate10 + kLossRate10Margin,
kPacketLossRate20 + kLossRate20Margin - 0.01, kPacketLossRate20 + kLossRate20Margin - 0.01,
kPacketLossRate10); kPacketLossRate10);
TestSetPacketLossRate(kPacketLossRate20 + kLossRate20Margin, TestSetPacketLossRate(encoder_.get(), kPacketLossRate20 + kLossRate20Margin,
1.0, 1.0, kPacketLossRate20);
kPacketLossRate20); TestSetPacketLossRate(encoder_.get(), kPacketLossRate20 + kLossRate20Margin,
TestSetPacketLossRate(kPacketLossRate20 + kLossRate20Margin,
kPacketLossRate20 - kLossRate20Margin, kPacketLossRate20 - kLossRate20Margin,
kPacketLossRate20); kPacketLossRate20);
TestSetPacketLossRate(kPacketLossRate20 - kLossRate20Margin - 0.01, TestSetPacketLossRate(
kPacketLossRate10 - kLossRate10Margin, encoder_.get(), kPacketLossRate20 - kLossRate20Margin - 0.01,
kPacketLossRate10); kPacketLossRate10 - kLossRate10Margin, kPacketLossRate10);
TestSetPacketLossRate(kPacketLossRate10 - kLossRate10Margin - 0.01, TestSetPacketLossRate(encoder_.get(),
kPacketLossRate5 - kLossRate5Margin, kPacketLossRate10 - kLossRate10Margin - 0.01,
kPacketLossRate5); kPacketLossRate5 - kLossRate5Margin, kPacketLossRate5);
TestSetPacketLossRate(kPacketLossRate5 - kLossRate5Margin - 0.01, TestSetPacketLossRate(encoder_.get(),
kPacketLossRate1, kPacketLossRate5 - kLossRate5Margin - 0.01,
kPacketLossRate1); kPacketLossRate1, kPacketLossRate1);
TestSetPacketLossRate(0.0, 0.0, 0.0); TestSetPacketLossRate(encoder_.get(), 0.0, 0.0, 0.0);
} }
} // namespace webrtc } // namespace webrtc

1
webrtc/modules/modules.gyp
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@ -97,7 +97,6 @@
], ],
'sources': [ 'sources': [
'audio_coding/codecs/cng/audio_encoder_cng_unittest.cc', 'audio_coding/codecs/cng/audio_encoder_cng_unittest.cc',
'audio_coding/codecs/opus/audio_encoder_mutable_opus_test.cc',
'audio_coding/main/acm2/acm_receiver_unittest.cc', 'audio_coding/main/acm2/acm_receiver_unittest.cc',
'audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc', 'audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc',
'audio_coding/main/acm2/audio_coding_module_unittest.cc', 'audio_coding/main/acm2/audio_coding_module_unittest.cc',