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/*
* Copyright (c) 2012 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 "webrtc/modules/audio_processing/agc/agc.h"
#include "gmock/gmock.h"
#include "gtest/gtest.h"
#include "webrtc/modules/audio_processing/agc/test/test_utils.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/test/testsupport/fileutils.h"
using ::testing::_;
using ::testing::AllOf;
using ::testing::AtLeast;
using ::testing::Eq;
using ::testing::Gt;
using ::testing::InSequence;
using ::testing::Lt;
using ::testing::Mock;
using ::testing::SaveArg;
namespace webrtc {
namespace {
// The tested values depend on this assumed gain.
const int kMaxGain = 80;
MATCHER_P(GtPointee, p, "") { return arg > *p; }
MATCHER_P(LtPointee, p, "") { return arg < *p; }
class AgcChecker {
public:
MOCK_METHOD2(LevelChanged, void(int iterations, int level));
};
class AgcTest : public ::testing::Test {
protected:
AgcTest()
: agc_(),
checker_(),
mic_level_(128) {
}
// A gain of <= -100 will zero out the signal.
void RunAgc(int iterations, float gain_db) {
FILE* input_file = fopen(
test::ResourcePath("voice_engine/audio_long16", "pcm").c_str(), "rb");
ASSERT_TRUE(input_file != NULL);
AudioFrame frame;
frame.sample_rate_hz_ = 16000;
frame.num_channels_ = 1;
frame.samples_per_channel_ = frame.sample_rate_hz_ / 100;
const size_t length = frame.samples_per_channel_ * frame.num_channels_;
float gain = Db2Linear(gain_db);
if (gain_db <= -100) {
gain = 0;
}
for (int i = 0; i < iterations; ++i) {
ASSERT_EQ(length, fread(frame.data_, sizeof(int16_t), length,
input_file));
SimulateMic(kMaxGain, mic_level_, &frame);
ApplyGainLinear(gain, &frame);
ASSERT_GE(agc_.Process(frame), 0);
int mic_level = agc_.MicLevel();
if (mic_level != mic_level_) {
printf("mic_level=%d\n", mic_level);
checker_.LevelChanged(i, mic_level);
}
mic_level_ = mic_level;
}
fclose(input_file);
}
Agc agc_;
AgcChecker checker_;
// Stores mic level between multiple runs of RunAgc in one test.
int mic_level_;
};
TEST_F(AgcTest, UpwardsChangeIsLimited) {
{
InSequence seq;
EXPECT_CALL(checker_, LevelChanged(Lt(500), Eq(179))).Times(1);
EXPECT_CALL(checker_, LevelChanged(_, Gt(179))).Times(AtLeast(1));
}
RunAgc(1000, -40);
}
TEST_F(AgcTest, DownwardsChangeIsLimited) {
{
InSequence seq;
EXPECT_CALL(checker_, LevelChanged(Lt(500), Eq(77))).Times(1);
EXPECT_CALL(checker_, LevelChanged(_, Lt(77))).Times(AtLeast(1));
}
RunAgc(1000, 40);
}
TEST_F(AgcTest, MovesUpToMaxAndDownToMin) {
int last_level = 128;
EXPECT_CALL(checker_, LevelChanged(_, GtPointee(&last_level)))
.Times(AtLeast(2))
.WillRepeatedly(SaveArg<1>(&last_level));
RunAgc(1000, -30);
EXPECT_EQ(255, last_level);
Mock::VerifyAndClearExpectations(&checker_);
EXPECT_CALL(checker_, LevelChanged(_, LtPointee(&last_level)))
.Times(AtLeast(2))
.WillRepeatedly(SaveArg<1>(&last_level));
RunAgc(1000, 50);
EXPECT_EQ(1, last_level);
}
TEST_F(AgcTest, HandlesZeroSignal) {
int last_level = 128;
// Doesn't respond to a zero signal.
EXPECT_CALL(checker_, LevelChanged(_, _)).Times(0);
RunAgc(1000, -100);
Mock::VerifyAndClearExpectations(&checker_);
// Reacts as usual afterwards.
EXPECT_CALL(checker_, LevelChanged(_, GtPointee(&last_level)))
.Times(AtLeast(2))
.WillRepeatedly(SaveArg<1>(&last_level));
RunAgc(500, -20);
}
TEST_F(AgcTest, ReachesSteadyState) {
int last_level = 128;
EXPECT_CALL(checker_, LevelChanged(_, _))
.Times(AtLeast(2))
.WillRepeatedly(SaveArg<1>(&last_level));
RunAgc(1000, -20);
Mock::VerifyAndClearExpectations(&checker_);
// If the level changes, it should be in a narrow band around the previous
// adaptation.
EXPECT_CALL(checker_, LevelChanged(_,
AllOf(Gt(last_level * 0.95), Lt(last_level * 1.05))))
.Times(AtLeast(0));
RunAgc(1000, -20);
}
// TODO(ajm): Add this test; requires measuring the signal RMS.
TEST_F(AgcTest, AdaptsToCorrectRMS) {
}
} // namespace
} // namespace webrtc
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