blob: 0f5a811badf24f3a0dccb0c94bdd6556f322f13c [file] [log] [blame]
/*
* Copyright (c) 2014 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 <memory>
#include <vector>
#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "test/gtest.h"
#include "test/mock_audio_encoder.h"
using ::testing::Return;
using ::testing::_;
using ::testing::SetArgPointee;
using ::testing::InSequence;
using ::testing::Invoke;
using ::testing::MockFunction;
namespace webrtc {
namespace {
static const size_t kMaxNumSamples = 48 * 10 * 2; // 10 ms @ 48 kHz stereo.
}
class AudioEncoderCopyRedTest : public ::testing::Test {
protected:
AudioEncoderCopyRedTest()
: mock_encoder_(new MockAudioEncoder),
timestamp_(4711),
sample_rate_hz_(16000),
num_audio_samples_10ms(sample_rate_hz_ / 100),
red_payload_type_(200) {
AudioEncoderCopyRed::Config config;
config.payload_type = red_payload_type_;
config.speech_encoder = std::unique_ptr<AudioEncoder>(mock_encoder_);
red_.reset(new AudioEncoderCopyRed(std::move(config)));
memset(audio_, 0, sizeof(audio_));
EXPECT_CALL(*mock_encoder_, NumChannels()).WillRepeatedly(Return(1U));
EXPECT_CALL(*mock_encoder_, SampleRateHz())
.WillRepeatedly(Return(sample_rate_hz_));
}
void TearDown() override { red_.reset(); }
void Encode() {
ASSERT_TRUE(red_.get() != NULL);
encoded_.Clear();
encoded_info_ = red_->Encode(
timestamp_,
rtc::ArrayView<const int16_t>(audio_, num_audio_samples_10ms),
&encoded_);
timestamp_ += rtc::checked_cast<uint32_t>(num_audio_samples_10ms);
}
MockAudioEncoder* mock_encoder_;
std::unique_ptr<AudioEncoderCopyRed> red_;
uint32_t timestamp_;
int16_t audio_[kMaxNumSamples];
const int sample_rate_hz_;
size_t num_audio_samples_10ms;
rtc::Buffer encoded_;
AudioEncoder::EncodedInfo encoded_info_;
const int red_payload_type_;
};
TEST_F(AudioEncoderCopyRedTest, CreateAndDestroy) {}
TEST_F(AudioEncoderCopyRedTest, CheckSampleRatePropagation) {
EXPECT_CALL(*mock_encoder_, SampleRateHz()).WillOnce(Return(17));
EXPECT_EQ(17, red_->SampleRateHz());
}
TEST_F(AudioEncoderCopyRedTest, CheckNumChannelsPropagation) {
EXPECT_CALL(*mock_encoder_, NumChannels()).WillOnce(Return(17U));
EXPECT_EQ(17U, red_->NumChannels());
}
TEST_F(AudioEncoderCopyRedTest, CheckFrameSizePropagation) {
EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
.WillOnce(Return(17U));
EXPECT_EQ(17U, red_->Num10MsFramesInNextPacket());
}
TEST_F(AudioEncoderCopyRedTest, CheckMaxFrameSizePropagation) {
EXPECT_CALL(*mock_encoder_, Max10MsFramesInAPacket()).WillOnce(Return(17U));
EXPECT_EQ(17U, red_->Max10MsFramesInAPacket());
}
TEST_F(AudioEncoderCopyRedTest, CheckTargetAudioBitratePropagation) {
EXPECT_CALL(*mock_encoder_,
OnReceivedUplinkBandwidth(4711, absl::optional<int64_t>()));
red_->OnReceivedUplinkBandwidth(4711, absl::nullopt);
}
TEST_F(AudioEncoderCopyRedTest, CheckPacketLossFractionPropagation) {
EXPECT_CALL(*mock_encoder_, OnReceivedUplinkPacketLossFraction(0.5));
red_->OnReceivedUplinkPacketLossFraction(0.5);
}
// Checks that the an Encode() call is immediately propagated to the speech
// encoder.
TEST_F(AudioEncoderCopyRedTest, CheckImmediateEncode) {
// Interleaving the EXPECT_CALL sequence with expectations on the MockFunction
// check ensures that exactly one call to EncodeImpl happens in each
// Encode call.
InSequence s;
MockFunction<void(int check_point_id)> check;
for (int i = 1; i <= 6; ++i) {
EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
.WillRepeatedly(Return(AudioEncoder::EncodedInfo()));
EXPECT_CALL(check, Call(i));
Encode();
check.Call(i);
}
}
// Checks that no output is produced if the underlying codec doesn't emit any
// new data, even if the RED codec is loaded with a secondary encoding.
TEST_F(AudioEncoderCopyRedTest, CheckNoOutput) {
static const size_t kEncodedSize = 17;
{
InSequence s;
EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
.WillOnce(Invoke(MockAudioEncoder::FakeEncoding(kEncodedSize)))
.WillOnce(Invoke(MockAudioEncoder::FakeEncoding(0)))
.WillOnce(Invoke(MockAudioEncoder::FakeEncoding(kEncodedSize)));
}
// Start with one Encode() call that will produce output.
Encode();
// First call is a special case, since it does not include a secondary
// payload.
EXPECT_EQ(1u, encoded_info_.redundant.size());
EXPECT_EQ(kEncodedSize, encoded_info_.encoded_bytes);
// Next call to the speech encoder will not produce any output.
Encode();
EXPECT_EQ(0u, encoded_info_.encoded_bytes);
// Final call to the speech encoder will produce output.
Encode();
EXPECT_EQ(2 * kEncodedSize, encoded_info_.encoded_bytes);
ASSERT_EQ(2u, encoded_info_.redundant.size());
}
// Checks that the correct payload sizes are populated into the redundancy
// information.
TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes) {
// Let the mock encoder return payload sizes 1, 2, 3, ..., 10 for the sequence
// of calls.
static const int kNumPackets = 10;
InSequence s;
for (int encode_size = 1; encode_size <= kNumPackets; ++encode_size) {
EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
.WillOnce(Invoke(MockAudioEncoder::FakeEncoding(encode_size)));
}
// First call is a special case, since it does not include a secondary
// payload.
Encode();
EXPECT_EQ(1u, encoded_info_.redundant.size());
EXPECT_EQ(1u, encoded_info_.encoded_bytes);
for (size_t i = 2; i <= kNumPackets; ++i) {
Encode();
ASSERT_EQ(2u, encoded_info_.redundant.size());
EXPECT_EQ(i, encoded_info_.redundant[0].encoded_bytes);
EXPECT_EQ(i - 1, encoded_info_.redundant[1].encoded_bytes);
EXPECT_EQ(i + i - 1, encoded_info_.encoded_bytes);
}
}
// Checks that the correct timestamps are returned.
TEST_F(AudioEncoderCopyRedTest, CheckTimestamps) {
uint32_t primary_timestamp = timestamp_;
AudioEncoder::EncodedInfo info;
info.encoded_bytes = 17;
info.encoded_timestamp = timestamp_;
EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
.WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
// First call is a special case, since it does not include a secondary
// payload.
Encode();
EXPECT_EQ(primary_timestamp, encoded_info_.encoded_timestamp);
uint32_t secondary_timestamp = primary_timestamp;
primary_timestamp = timestamp_;
info.encoded_timestamp = timestamp_;
EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
.WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
Encode();
ASSERT_EQ(2u, encoded_info_.redundant.size());
EXPECT_EQ(primary_timestamp, encoded_info_.redundant[0].encoded_timestamp);
EXPECT_EQ(secondary_timestamp, encoded_info_.redundant[1].encoded_timestamp);
EXPECT_EQ(primary_timestamp, encoded_info_.encoded_timestamp);
}
// Checks that the primary and secondary payloads are written correctly.
TEST_F(AudioEncoderCopyRedTest, CheckPayloads) {
// Let the mock encoder write payloads with increasing values. The first
// payload will have values 0, 1, 2, ..., kPayloadLenBytes - 1.
static const size_t kPayloadLenBytes = 5;
uint8_t payload[kPayloadLenBytes];
for (uint8_t i = 0; i < kPayloadLenBytes; ++i) {
payload[i] = i;
}
EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
.WillRepeatedly(Invoke(MockAudioEncoder::CopyEncoding(payload)));
// First call is a special case, since it does not include a secondary
// payload.
Encode();
EXPECT_EQ(kPayloadLenBytes, encoded_info_.encoded_bytes);
for (size_t i = 0; i < kPayloadLenBytes; ++i) {
EXPECT_EQ(i, encoded_.data()[i]);
}
for (int j = 0; j < 5; ++j) {
// Increment all values of the payload by 10.
for (size_t i = 0; i < kPayloadLenBytes; ++i)
payload[i] += 10;
Encode();
ASSERT_EQ(2u, encoded_info_.redundant.size());
EXPECT_EQ(kPayloadLenBytes, encoded_info_.redundant[0].encoded_bytes);
EXPECT_EQ(kPayloadLenBytes, encoded_info_.redundant[1].encoded_bytes);
for (size_t i = 0; i < kPayloadLenBytes; ++i) {
// Check primary payload.
EXPECT_EQ((j + 1) * 10 + i, encoded_.data()[i]);
// Check secondary payload.
EXPECT_EQ(j * 10 + i, encoded_.data()[i + kPayloadLenBytes]);
}
}
}
// Checks correct propagation of payload type.
// Checks that the correct timestamps are returned.
TEST_F(AudioEncoderCopyRedTest, CheckPayloadType) {
const int primary_payload_type = red_payload_type_ + 1;
AudioEncoder::EncodedInfo info;
info.encoded_bytes = 17;
info.payload_type = primary_payload_type;
EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
.WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
// First call is a special case, since it does not include a secondary
// payload.
Encode();
ASSERT_EQ(1u, encoded_info_.redundant.size());
EXPECT_EQ(primary_payload_type, encoded_info_.redundant[0].payload_type);
EXPECT_EQ(red_payload_type_, encoded_info_.payload_type);
const int secondary_payload_type = red_payload_type_ + 2;
info.payload_type = secondary_payload_type;
EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
.WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
Encode();
ASSERT_EQ(2u, encoded_info_.redundant.size());
EXPECT_EQ(secondary_payload_type, encoded_info_.redundant[0].payload_type);
EXPECT_EQ(primary_payload_type, encoded_info_.redundant[1].payload_type);
EXPECT_EQ(red_payload_type_, encoded_info_.payload_type);
}
#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// This test fixture tests various error conditions that makes the
// AudioEncoderCng die via CHECKs.
class AudioEncoderCopyRedDeathTest : public AudioEncoderCopyRedTest {
protected:
AudioEncoderCopyRedDeathTest() : AudioEncoderCopyRedTest() {}
};
TEST_F(AudioEncoderCopyRedDeathTest, WrongFrameSize) {
num_audio_samples_10ms *= 2; // 20 ms frame.
EXPECT_DEATH(Encode(), "");
num_audio_samples_10ms = 0; // Zero samples.
EXPECT_DEATH(Encode(), "");
}
TEST_F(AudioEncoderCopyRedDeathTest, NullSpeechEncoder) {
AudioEncoderCopyRed* red = NULL;
AudioEncoderCopyRed::Config config;
config.speech_encoder = NULL;
EXPECT_DEATH(red = new AudioEncoderCopyRed(std::move(config)),
"Speech encoder not provided.");
// The delete operation is needed to avoid leak reports from memcheck.
delete red;
}
#endif // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
} // namespace webrtc