mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-13 22:00:47 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
Jakob Ivarsson 918eb19303 Fix crash when Opus maxptime < 20ms. 
A follow up cl will be created to better handle nullopt frame length range in AudioSendStream.

Note that maxptime is still not used for setting the frame length (only ptime is).

Bug: chromium:1109337
Change-Id: Id21fd8c76a6c4a0c85719a955116f8d16001a3d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284501
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38702}
2022-11-22 01:21:24 +00:00

917 lines
36 KiB
C++
Raw Blame History

/*
* 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.
*/
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include <array>
#include <memory>
#include <utility>
#include "absl/strings/string_view.h"
#include "common_audio/mocks/mock_smoothing_filter.h"
#include "modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h"
#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
#include "modules/audio_coding/neteq/tools/audio_loop.h"
#include "rtc_base/checks.h"
#include "rtc_base/fake_clock.h"
#include "test/field_trial.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
using ::testing::NiceMock;
using ::testing::Return;
namespace {
constexpr int kDefaultOpusPayloadType = 105;
constexpr int kDefaultOpusRate = 32000;
constexpr int kDefaultOpusPacSize = 960;
constexpr int64_t kInitialTimeUs = 12345678;
AudioEncoderOpusConfig CreateConfigWithParameters(
const SdpAudioFormat::Parameters& params) {
const SdpAudioFormat format("opus", 48000, 2, params);
return *AudioEncoderOpus::SdpToConfig(format);
}
struct AudioEncoderOpusStates {
MockAudioNetworkAdaptor* mock_audio_network_adaptor;
MockSmoothingFilter* mock_bitrate_smoother;
std::unique_ptr<AudioEncoderOpusImpl> encoder;
std::unique_ptr<rtc::ScopedFakeClock> fake_clock;
AudioEncoderOpusConfig config;
};
std::unique_ptr<AudioEncoderOpusStates> CreateCodec(int sample_rate_hz,
size_t num_channels) {
std::unique_ptr<AudioEncoderOpusStates> states =
std::make_unique<AudioEncoderOpusStates>();
states->mock_audio_network_adaptor = nullptr;
states->fake_clock.reset(new rtc::ScopedFakeClock());
states->fake_clock->SetTime(Timestamp::Micros(kInitialTimeUs));
MockAudioNetworkAdaptor** mock_ptr = &states->mock_audio_network_adaptor;
AudioEncoderOpusImpl::AudioNetworkAdaptorCreator creator =
[mock_ptr](absl::string_view, RtcEventLog* event_log) {
std::unique_ptr<MockAudioNetworkAdaptor> adaptor(
new NiceMock<MockAudioNetworkAdaptor>());
EXPECT_CALL(*adaptor, Die());
*mock_ptr = adaptor.get();
return adaptor;
};
AudioEncoderOpusConfig config;
config.frame_size_ms = rtc::CheckedDivExact(kDefaultOpusPacSize, 48);
config.sample_rate_hz = sample_rate_hz;
config.num_channels = num_channels;
config.bitrate_bps = kDefaultOpusRate;
config.application = num_channels == 1
? AudioEncoderOpusConfig::ApplicationMode::kVoip
: AudioEncoderOpusConfig::ApplicationMode::kAudio;
config.supported_frame_lengths_ms.push_back(config.frame_size_ms);
states->config = config;
std::unique_ptr<MockSmoothingFilter> bitrate_smoother(
new MockSmoothingFilter());
states->mock_bitrate_smoother = bitrate_smoother.get();
states->encoder.reset(
new AudioEncoderOpusImpl(states->config, kDefaultOpusPayloadType, creator,
std::move(bitrate_smoother)));
return states;
}
AudioEncoderRuntimeConfig CreateEncoderRuntimeConfig() {
constexpr int kBitrate = 40000;
constexpr int kFrameLength = 60;
constexpr bool kEnableDtx = false;
constexpr size_t kNumChannels = 1;
AudioEncoderRuntimeConfig config;
config.bitrate_bps = kBitrate;
config.frame_length_ms = kFrameLength;
config.enable_dtx = kEnableDtx;
config.num_channels = kNumChannels;
return config;
}
void CheckEncoderRuntimeConfig(const AudioEncoderOpusImpl* encoder,
const AudioEncoderRuntimeConfig& config) {
EXPECT_EQ(*config.bitrate_bps, encoder->GetTargetBitrate());
EXPECT_EQ(*config.frame_length_ms, encoder->next_frame_length_ms());
EXPECT_EQ(*config.enable_dtx, encoder->GetDtx());
EXPECT_EQ(*config.num_channels, encoder->num_channels_to_encode());
}
// Create 10ms audio data blocks for a total packet size of "packet_size_ms".
std::unique_ptr<test::AudioLoop> Create10msAudioBlocks(
const std::unique_ptr<AudioEncoderOpusImpl>& encoder,
int packet_size_ms) {
const std::string file_name =
test::ResourcePath("audio_coding/testfile32kHz", "pcm");
std::unique_ptr<test::AudioLoop> speech_data(new test::AudioLoop());
int audio_samples_per_ms =
rtc::CheckedDivExact(encoder->SampleRateHz(), 1000);
if (!speech_data->Init(
file_name,
packet_size_ms * audio_samples_per_ms *
encoder->num_channels_to_encode(),
10 * audio_samples_per_ms * encoder->num_channels_to_encode()))
return nullptr;
return speech_data;
}
} // namespace
class AudioEncoderOpusTest : public ::testing::TestWithParam<int> {
protected:
int sample_rate_hz_{GetParam()};
};
INSTANTIATE_TEST_SUITE_P(Param,
AudioEncoderOpusTest,
::testing::Values(16000, 48000));
TEST_P(AudioEncoderOpusTest, DefaultApplicationModeMono) {
auto states = CreateCodec(sample_rate_hz_, 1);
EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
states->encoder->application());
}
TEST_P(AudioEncoderOpusTest, DefaultApplicationModeStereo) {
auto states = CreateCodec(sample_rate_hz_, 2);
EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kAudio,
states->encoder->application());
}
TEST_P(AudioEncoderOpusTest, ChangeApplicationMode) {
auto states = CreateCodec(sample_rate_hz_, 2);
EXPECT_TRUE(
states->encoder->SetApplication(AudioEncoder::Application::kSpeech));
EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
states->encoder->application());
}
TEST_P(AudioEncoderOpusTest, ResetWontChangeApplicationMode) {
auto states = CreateCodec(sample_rate_hz_, 2);
// Trigger a reset.
states->encoder->Reset();
// Verify that the mode is still kAudio.
EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kAudio,
states->encoder->application());
// Now change to kVoip.
EXPECT_TRUE(
states->encoder->SetApplication(AudioEncoder::Application::kSpeech));
EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
states->encoder->application());
// Trigger a reset again.
states->encoder->Reset();
// Verify that the mode is still kVoip.
EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
states->encoder->application());
}
TEST_P(AudioEncoderOpusTest, ToggleDtx) {
auto states = CreateCodec(sample_rate_hz_, 2);
// Enable DTX
EXPECT_TRUE(states->encoder->SetDtx(true));
EXPECT_TRUE(states->encoder->GetDtx());
// Turn off DTX.
EXPECT_TRUE(states->encoder->SetDtx(false));
EXPECT_FALSE(states->encoder->GetDtx());
}
TEST_P(AudioEncoderOpusTest,
OnReceivedUplinkBandwidthWithoutAudioNetworkAdaptor) {
auto states = CreateCodec(sample_rate_hz_, 1);
// Constants are replicated from audio_states->encoderopus.cc.
const int kMinBitrateBps = 6000;
const int kMaxBitrateBps = 510000;
const int kOverheadBytesPerPacket = 64;
states->encoder->OnReceivedOverhead(kOverheadBytesPerPacket);
const int kOverheadBps = 8 * kOverheadBytesPerPacket *
rtc::CheckedDivExact(48000, kDefaultOpusPacSize);
// Set a too low bitrate.
states->encoder->OnReceivedUplinkBandwidth(kMinBitrateBps + kOverheadBps - 1,
absl::nullopt);
EXPECT_EQ(kMinBitrateBps, states->encoder->GetTargetBitrate());
// Set a too high bitrate.
states->encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps + kOverheadBps + 1,
absl::nullopt);
EXPECT_EQ(kMaxBitrateBps, states->encoder->GetTargetBitrate());
// Set the minimum rate.
states->encoder->OnReceivedUplinkBandwidth(kMinBitrateBps + kOverheadBps,
absl::nullopt);
EXPECT_EQ(kMinBitrateBps, states->encoder->GetTargetBitrate());
// Set the maximum rate.
states->encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps + kOverheadBps,
absl::nullopt);
EXPECT_EQ(kMaxBitrateBps, states->encoder->GetTargetBitrate());
// Set rates from kMaxBitrateBps up to 32000 bps.
for (int rate = kMinBitrateBps + kOverheadBps; rate <= 32000 + kOverheadBps;
rate += 1000) {
states->encoder->OnReceivedUplinkBandwidth(rate, absl::nullopt);
EXPECT_EQ(rate - kOverheadBps, states->encoder->GetTargetBitrate());
}
}
TEST_P(AudioEncoderOpusTest, SetReceiverFrameLengthRange) {
auto states = CreateCodec(sample_rate_hz_, 2);
// Before calling to `SetReceiverFrameLengthRange`,
// `supported_frame_lengths_ms` should contain only the frame length being
// used.
using ::testing::ElementsAre;
EXPECT_THAT(states->encoder->supported_frame_lengths_ms(),
ElementsAre(states->encoder->next_frame_length_ms()));
states->encoder->SetReceiverFrameLengthRange(0, 12345);
states->encoder->SetReceiverFrameLengthRange(21, 60);
EXPECT_THAT(states->encoder->supported_frame_lengths_ms(),
ElementsAre(40, 60));
states->encoder->SetReceiverFrameLengthRange(20, 59);
EXPECT_THAT(states->encoder->supported_frame_lengths_ms(),
ElementsAre(20, 40));
}
TEST_P(AudioEncoderOpusTest,
InvokeAudioNetworkAdaptorOnReceivedUplinkPacketLossFraction) {
auto states = CreateCodec(sample_rate_hz_, 2);
states->encoder->EnableAudioNetworkAdaptor("", nullptr);
auto config = CreateEncoderRuntimeConfig();
EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
.WillOnce(Return(config));
// Since using mock audio network adaptor, any packet loss fraction is fine.
constexpr float kUplinkPacketLoss = 0.1f;
EXPECT_CALL(*states->mock_audio_network_adaptor,
SetUplinkPacketLossFraction(kUplinkPacketLoss));
states->encoder->OnReceivedUplinkPacketLossFraction(kUplinkPacketLoss);
CheckEncoderRuntimeConfig(states->encoder.get(), config);
}
TEST_P(AudioEncoderOpusTest,
InvokeAudioNetworkAdaptorOnReceivedUplinkBandwidth) {
test::ScopedFieldTrials override_field_trials(
"WebRTC-Audio-StableTargetAdaptation/Disabled/");
auto states = CreateCodec(sample_rate_hz_, 2);
states->encoder->EnableAudioNetworkAdaptor("", nullptr);
auto config = CreateEncoderRuntimeConfig();
EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
.WillOnce(Return(config));
// Since using mock audio network adaptor, any target audio bitrate is fine.
constexpr int kTargetAudioBitrate = 30000;
constexpr int64_t kProbingIntervalMs = 3000;
EXPECT_CALL(*states->mock_audio_network_adaptor,
SetTargetAudioBitrate(kTargetAudioBitrate));
EXPECT_CALL(*states->mock_bitrate_smoother,
SetTimeConstantMs(kProbingIntervalMs * 4));
EXPECT_CALL(*states->mock_bitrate_smoother, AddSample(kTargetAudioBitrate));
states->encoder->OnReceivedUplinkBandwidth(kTargetAudioBitrate,
kProbingIntervalMs);
CheckEncoderRuntimeConfig(states->encoder.get(), config);
}
TEST_P(AudioEncoderOpusTest,
InvokeAudioNetworkAdaptorOnReceivedUplinkAllocation) {
auto states = CreateCodec(sample_rate_hz_, 2);
states->encoder->EnableAudioNetworkAdaptor("", nullptr);
auto config = CreateEncoderRuntimeConfig();
EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
.WillOnce(Return(config));
BitrateAllocationUpdate update;
update.target_bitrate = DataRate::BitsPerSec(30000);
update.stable_target_bitrate = DataRate::BitsPerSec(20000);
update.bwe_period = TimeDelta::Millis(200);
EXPECT_CALL(*states->mock_audio_network_adaptor,
SetTargetAudioBitrate(update.target_bitrate.bps()));
EXPECT_CALL(*states->mock_audio_network_adaptor,
SetUplinkBandwidth(update.stable_target_bitrate.bps()));
states->encoder->OnReceivedUplinkAllocation(update);
CheckEncoderRuntimeConfig(states->encoder.get(), config);
}
TEST_P(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedRtt) {
auto states = CreateCodec(sample_rate_hz_, 2);
states->encoder->EnableAudioNetworkAdaptor("", nullptr);
auto config = CreateEncoderRuntimeConfig();
EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
.WillOnce(Return(config));
// Since using mock audio network adaptor, any rtt is fine.
constexpr int kRtt = 30;
EXPECT_CALL(*states->mock_audio_network_adaptor, SetRtt(kRtt));
states->encoder->OnReceivedRtt(kRtt);
CheckEncoderRuntimeConfig(states->encoder.get(), config);
}
TEST_P(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedOverhead) {
auto states = CreateCodec(sample_rate_hz_, 2);
states->encoder->EnableAudioNetworkAdaptor("", nullptr);
auto config = CreateEncoderRuntimeConfig();
EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
.WillOnce(Return(config));
// Since using mock audio network adaptor, any overhead is fine.
constexpr size_t kOverhead = 64;
EXPECT_CALL(*states->mock_audio_network_adaptor, SetOverhead(kOverhead));
states->encoder->OnReceivedOverhead(kOverhead);
CheckEncoderRuntimeConfig(states->encoder.get(), config);
}
TEST_P(AudioEncoderOpusTest,
PacketLossFractionSmoothedOnSetUplinkPacketLossFraction) {
auto states = CreateCodec(sample_rate_hz_, 2);
// The values are carefully chosen so that if no smoothing is made, the test
// will fail.
constexpr float kPacketLossFraction_1 = 0.02f;
constexpr float kPacketLossFraction_2 = 0.198f;
// `kSecondSampleTimeMs` is chosen to ease the calculation since
// 0.9999 ^ 6931 = 0.5.
constexpr int64_t kSecondSampleTimeMs = 6931;
// First time, no filtering.
states->encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_1);
EXPECT_FLOAT_EQ(0.02f, states->encoder->packet_loss_rate());
states->fake_clock->AdvanceTime(TimeDelta::Millis(kSecondSampleTimeMs));
states->encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_2);
// Now the output of packet loss fraction smoother should be
// (0.02 + 0.198) / 2 = 0.109.
EXPECT_NEAR(0.109f, states->encoder->packet_loss_rate(), 0.001);
}
TEST_P(AudioEncoderOpusTest, PacketLossRateUpperBounded) {
auto states = CreateCodec(sample_rate_hz_, 2);
states->encoder->OnReceivedUplinkPacketLossFraction(0.5);
EXPECT_FLOAT_EQ(0.2f, states->encoder->packet_loss_rate());
}
TEST_P(AudioEncoderOpusTest, DoNotInvokeSetTargetBitrateIfOverheadUnknown) {
test::ScopedFieldTrials override_field_trials(
"WebRTC-SendSideBwe-WithOverhead/Enabled/");
auto states = CreateCodec(sample_rate_hz_, 2);
states->encoder->OnReceivedUplinkBandwidth(kDefaultOpusRate * 2,
absl::nullopt);
// Since `OnReceivedOverhead` has not been called, the codec bitrate should
// not change.
EXPECT_EQ(kDefaultOpusRate, states->encoder->GetTargetBitrate());
}
// Verifies that the complexity adaptation in the config works as intended.
TEST(AudioEncoderOpusTest, ConfigComplexityAdaptation) {
AudioEncoderOpusConfig config;
config.low_rate_complexity = 8;
config.complexity = 6;
// Bitrate within hysteresis window. Expect empty output.
config.bitrate_bps = 12500;
EXPECT_EQ(absl::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
// Bitrate below hysteresis window. Expect higher complexity.
config.bitrate_bps = 10999;
EXPECT_EQ(8, AudioEncoderOpusImpl::GetNewComplexity(config));
// Bitrate within hysteresis window. Expect empty output.
config.bitrate_bps = 12500;
EXPECT_EQ(absl::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
// Bitrate above hysteresis window. Expect lower complexity.
config.bitrate_bps = 14001;
EXPECT_EQ(6, AudioEncoderOpusImpl::GetNewComplexity(config));
}
// Verifies that the bandwidth adaptation in the config works as intended.
TEST_P(AudioEncoderOpusTest, ConfigBandwidthAdaptation) {
AudioEncoderOpusConfig config;
const size_t opus_rate_khz = rtc::CheckedDivExact(sample_rate_hz_, 1000);
const std::vector<int16_t> silence(
opus_rate_khz * config.frame_size_ms * config.num_channels, 0);
constexpr size_t kMaxBytes = 1000;
uint8_t bitstream[kMaxBytes];
OpusEncInst* inst;
EXPECT_EQ(0, WebRtcOpus_EncoderCreate(
&inst, config.num_channels,
config.application ==
AudioEncoderOpusConfig::ApplicationMode::kVoip
? 0
: 1,
sample_rate_hz_));
// Bitrate below minmum wideband. Expect narrowband.
config.bitrate_bps = absl::optional<int>(7999);
auto bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
EXPECT_EQ(absl::optional<int>(OPUS_BANDWIDTH_NARROWBAND), bandwidth);
WebRtcOpus_SetBandwidth(inst, *bandwidth);
// It is necessary to encode here because Opus has some logic in the encoder
// that goes from the user-set bandwidth to the used and returned one.
WebRtcOpus_Encode(inst, silence.data(),
rtc::CheckedDivExact(silence.size(), config.num_channels),
kMaxBytes, bitstream);
// Bitrate not yet above maximum narrowband. Expect empty.
config.bitrate_bps = absl::optional<int>(9000);
bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
EXPECT_EQ(absl::optional<int>(), bandwidth);
// Bitrate above maximum narrowband. Expect wideband.
config.bitrate_bps = absl::optional<int>(9001);
bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
EXPECT_EQ(absl::optional<int>(OPUS_BANDWIDTH_WIDEBAND), bandwidth);
WebRtcOpus_SetBandwidth(inst, *bandwidth);
// It is necessary to encode here because Opus has some logic in the encoder
// that goes from the user-set bandwidth to the used and returned one.
WebRtcOpus_Encode(inst, silence.data(),
rtc::CheckedDivExact(silence.size(), config.num_channels),
kMaxBytes, bitstream);
// Bitrate not yet below minimum wideband. Expect empty.
config.bitrate_bps = absl::optional<int>(8000);
bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
EXPECT_EQ(absl::optional<int>(), bandwidth);
// Bitrate above automatic threshold. Expect automatic.
config.bitrate_bps = absl::optional<int>(12001);
bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
EXPECT_EQ(absl::optional<int>(OPUS_AUTO), bandwidth);
EXPECT_EQ(0, WebRtcOpus_EncoderFree(inst));
}
TEST_P(AudioEncoderOpusTest, EmptyConfigDoesNotAffectEncoderSettings) {
auto states = CreateCodec(sample_rate_hz_, 2);
states->encoder->EnableAudioNetworkAdaptor("", nullptr);
auto config = CreateEncoderRuntimeConfig();
AudioEncoderRuntimeConfig empty_config;
EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
.WillOnce(Return(config))
.WillOnce(Return(empty_config));
constexpr size_t kOverhead = 64;
EXPECT_CALL(*states->mock_audio_network_adaptor, SetOverhead(kOverhead))
.Times(2);
states->encoder->OnReceivedOverhead(kOverhead);
states->encoder->OnReceivedOverhead(kOverhead);
CheckEncoderRuntimeConfig(states->encoder.get(), config);
}
TEST_P(AudioEncoderOpusTest, UpdateUplinkBandwidthInAudioNetworkAdaptor) {
test::ScopedFieldTrials override_field_trials(
"WebRTC-Audio-StableTargetAdaptation/Disabled/");
auto states = CreateCodec(sample_rate_hz_, 2);
states->encoder->EnableAudioNetworkAdaptor("", nullptr);
const size_t opus_rate_khz = rtc::CheckedDivExact(sample_rate_hz_, 1000);
const std::vector<int16_t> audio(opus_rate_khz * 10 * 2, 0);
rtc::Buffer encoded;
EXPECT_CALL(*states->mock_bitrate_smoother, GetAverage())
.WillOnce(Return(50000));
EXPECT_CALL(*states->mock_audio_network_adaptor, SetUplinkBandwidth(50000));
states->encoder->Encode(
0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
// Repeat update uplink bandwidth tests.
for (int i = 0; i < 5; i++) {
// Don't update till it is time to update again.
states->fake_clock->AdvanceTime(TimeDelta::Millis(
states->config.uplink_bandwidth_update_interval_ms - 1));
states->encoder->Encode(
0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
// Update when it is time to update.
EXPECT_CALL(*states->mock_bitrate_smoother, GetAverage())
.WillOnce(Return(40000));
EXPECT_CALL(*states->mock_audio_network_adaptor, SetUplinkBandwidth(40000));
states->fake_clock->AdvanceTime(TimeDelta::Millis(1));
states->encoder->Encode(
0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
}
}
TEST_P(AudioEncoderOpusTest, EncodeAtMinBitrate) {
auto states = CreateCodec(sample_rate_hz_, 1);
constexpr int kNumPacketsToEncode = 2;
auto audio_frames =
Create10msAudioBlocks(states->encoder, kNumPacketsToEncode * 20);
ASSERT_TRUE(audio_frames) << "Create10msAudioBlocks failed";
rtc::Buffer encoded;
uint32_t rtp_timestamp = 12345; // Just a number not important to this test.
states->encoder->OnReceivedUplinkBandwidth(0, absl::nullopt);
for (int packet_index = 0; packet_index < kNumPacketsToEncode;
packet_index++) {
// Make sure we are not encoding before we have enough data for
// a 20ms packet.
for (int index = 0; index < 1; index++) {
states->encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
&encoded);
EXPECT_EQ(0u, encoded.size());
}
// Should encode now.
states->encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
&encoded);
EXPECT_GT(encoded.size(), 0u);
encoded.Clear();
}
}
TEST(AudioEncoderOpusTest, TestConfigDefaults) {
const auto config_opt = AudioEncoderOpus::SdpToConfig({"opus", 48000, 2});
ASSERT_TRUE(config_opt);
EXPECT_EQ(48000, config_opt->max_playback_rate_hz);
EXPECT_EQ(1u, config_opt->num_channels);
EXPECT_FALSE(config_opt->fec_enabled);
EXPECT_FALSE(config_opt->dtx_enabled);
EXPECT_EQ(20, config_opt->frame_size_ms);
}
TEST(AudioEncoderOpusTest, TestConfigFromParams) {
const auto config1 = CreateConfigWithParameters({{"stereo", "0"}});
EXPECT_EQ(1U, config1.num_channels);
const auto config2 = CreateConfigWithParameters({{"stereo", "1"}});
EXPECT_EQ(2U, config2.num_channels);
const auto config3 = CreateConfigWithParameters({{"useinbandfec", "0"}});
EXPECT_FALSE(config3.fec_enabled);
const auto config4 = CreateConfigWithParameters({{"useinbandfec", "1"}});
EXPECT_TRUE(config4.fec_enabled);
const auto config5 = CreateConfigWithParameters({{"usedtx", "0"}});
EXPECT_FALSE(config5.dtx_enabled);
const auto config6 = CreateConfigWithParameters({{"usedtx", "1"}});
EXPECT_TRUE(config6.dtx_enabled);
const auto config7 = CreateConfigWithParameters({{"cbr", "0"}});
EXPECT_FALSE(config7.cbr_enabled);
const auto config8 = CreateConfigWithParameters({{"cbr", "1"}});
EXPECT_TRUE(config8.cbr_enabled);
const auto config9 =
CreateConfigWithParameters({{"maxplaybackrate", "12345"}});
EXPECT_EQ(12345, config9.max_playback_rate_hz);
const auto config10 =
CreateConfigWithParameters({{"maxaveragebitrate", "96000"}});
EXPECT_EQ(96000, config10.bitrate_bps);
const auto config11 = CreateConfigWithParameters({{"maxptime", "40"}});
for (int frame_length : config11.supported_frame_lengths_ms) {
EXPECT_LE(frame_length, 40);
}
const auto config12 = CreateConfigWithParameters({{"minptime", "40"}});
for (int frame_length : config12.supported_frame_lengths_ms) {
EXPECT_GE(frame_length, 40);
}
const auto config13 = CreateConfigWithParameters({{"ptime", "40"}});
EXPECT_EQ(40, config13.frame_size_ms);
constexpr int kMinSupportedFrameLength = 10;
constexpr int kMaxSupportedFrameLength =
WEBRTC_OPUS_SUPPORT_120MS_PTIME ? 120 : 60;
const auto config14 = CreateConfigWithParameters({{"ptime", "1"}});
EXPECT_EQ(kMinSupportedFrameLength, config14.frame_size_ms);
const auto config15 = CreateConfigWithParameters({{"ptime", "2000"}});
EXPECT_EQ(kMaxSupportedFrameLength, config15.frame_size_ms);
}
TEST(AudioEncoderOpusTest, TestConfigFromInvalidParams) {
const webrtc::SdpAudioFormat format("opus", 48000, 2);
const auto default_config = *AudioEncoderOpus::SdpToConfig(format);
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
const std::vector<int> default_supported_frame_lengths_ms({20, 40, 60, 120});
#else
const std::vector<int> default_supported_frame_lengths_ms({20, 40, 60});
#endif
AudioEncoderOpusConfig config;
config = CreateConfigWithParameters({{"stereo", "invalid"}});
EXPECT_EQ(default_config.num_channels, config.num_channels);
config = CreateConfigWithParameters({{"useinbandfec", "invalid"}});
EXPECT_EQ(default_config.fec_enabled, config.fec_enabled);
config = CreateConfigWithParameters({{"usedtx", "invalid"}});
EXPECT_EQ(default_config.dtx_enabled, config.dtx_enabled);
config = CreateConfigWithParameters({{"cbr", "invalid"}});
EXPECT_EQ(default_config.dtx_enabled, config.dtx_enabled);
config = CreateConfigWithParameters({{"maxplaybackrate", "0"}});
EXPECT_EQ(default_config.max_playback_rate_hz, config.max_playback_rate_hz);
config = CreateConfigWithParameters({{"maxplaybackrate", "-23"}});
EXPECT_EQ(default_config.max_playback_rate_hz, config.max_playback_rate_hz);
config = CreateConfigWithParameters({{"maxplaybackrate", "not a number!"}});
EXPECT_EQ(default_config.max_playback_rate_hz, config.max_playback_rate_hz);
config = CreateConfigWithParameters({{"maxaveragebitrate", "0"}});
EXPECT_EQ(6000, config.bitrate_bps);
config = CreateConfigWithParameters({{"maxaveragebitrate", "-1000"}});
EXPECT_EQ(6000, config.bitrate_bps);
config = CreateConfigWithParameters({{"maxaveragebitrate", "1024000"}});
EXPECT_EQ(510000, config.bitrate_bps);
config = CreateConfigWithParameters({{"maxaveragebitrate", "not a number!"}});
EXPECT_EQ(default_config.bitrate_bps, config.bitrate_bps);
config = CreateConfigWithParameters({{"maxptime", "invalid"}});
EXPECT_EQ(default_supported_frame_lengths_ms,
config.supported_frame_lengths_ms);
config = CreateConfigWithParameters({{"minptime", "invalid"}});
EXPECT_EQ(default_supported_frame_lengths_ms,
config.supported_frame_lengths_ms);
config = CreateConfigWithParameters({{"ptime", "invalid"}});
EXPECT_EQ(default_supported_frame_lengths_ms,
config.supported_frame_lengths_ms);
}
TEST(AudioEncoderOpusTest, GetFrameLenghtRange) {
AudioEncoderOpusConfig config =
CreateConfigWithParameters({{"maxptime", "10"}, {"ptime", "10"}});
std::unique_ptr<AudioEncoder> encoder =
AudioEncoderOpus::MakeAudioEncoder(config, kDefaultOpusPayloadType);
auto ptime = webrtc::TimeDelta::Millis(10);
absl::optional<std::pair<webrtc::TimeDelta, webrtc::TimeDelta>> range = {
{ptime, ptime}};
EXPECT_EQ(encoder->GetFrameLengthRange(), range);
}
// Test that bitrate will be overridden by the "maxaveragebitrate" parameter.
// Also test that the "maxaveragebitrate" can't be set to values outside the
// range of 6000 and 510000
TEST(AudioEncoderOpusTest, SetSendCodecOpusMaxAverageBitrate) {
// Ignore if less than 6000.
const auto config1 = AudioEncoderOpus::SdpToConfig(
{"opus", 48000, 2, {{"maxaveragebitrate", "5999"}}});
EXPECT_EQ(6000, config1->bitrate_bps);
// Ignore if larger than 510000.
const auto config2 = AudioEncoderOpus::SdpToConfig(
{"opus", 48000, 2, {{"maxaveragebitrate", "510001"}}});
EXPECT_EQ(510000, config2->bitrate_bps);
const auto config3 = AudioEncoderOpus::SdpToConfig(
{"opus", 48000, 2, {{"maxaveragebitrate", "200000"}}});
EXPECT_EQ(200000, config3->bitrate_bps);
}
// Test maxplaybackrate <= 8000 triggers Opus narrow band mode.
TEST(AudioEncoderOpusTest, SetMaxPlaybackRateNb) {
auto config = CreateConfigWithParameters({{"maxplaybackrate", "8000"}});
EXPECT_EQ(8000, config.max_playback_rate_hz);
EXPECT_EQ(12000, config.bitrate_bps);
config = CreateConfigWithParameters(
{{"maxplaybackrate", "8000"}, {"stereo", "1"}});
EXPECT_EQ(8000, config.max_playback_rate_hz);
EXPECT_EQ(24000, config.bitrate_bps);
}
// Test 8000 < maxplaybackrate <= 12000 triggers Opus medium band mode.
TEST(AudioEncoderOpusTest, SetMaxPlaybackRateMb) {
auto config = CreateConfigWithParameters({{"maxplaybackrate", "8001"}});
EXPECT_EQ(8001, config.max_playback_rate_hz);
EXPECT_EQ(20000, config.bitrate_bps);
config = CreateConfigWithParameters(
{{"maxplaybackrate", "8001"}, {"stereo", "1"}});
EXPECT_EQ(8001, config.max_playback_rate_hz);
EXPECT_EQ(40000, config.bitrate_bps);
}
// Test 12000 < maxplaybackrate <= 16000 triggers Opus wide band mode.
TEST(AudioEncoderOpusTest, SetMaxPlaybackRateWb) {
auto config = CreateConfigWithParameters({{"maxplaybackrate", "12001"}});
EXPECT_EQ(12001, config.max_playback_rate_hz);
EXPECT_EQ(20000, config.bitrate_bps);
config = CreateConfigWithParameters(
{{"maxplaybackrate", "12001"}, {"stereo", "1"}});
EXPECT_EQ(12001, config.max_playback_rate_hz);
EXPECT_EQ(40000, config.bitrate_bps);
}
// Test 16000 < maxplaybackrate <= 24000 triggers Opus super wide band mode.
TEST(AudioEncoderOpusTest, SetMaxPlaybackRateSwb) {
auto config = CreateConfigWithParameters({{"maxplaybackrate", "16001"}});
EXPECT_EQ(16001, config.max_playback_rate_hz);
EXPECT_EQ(32000, config.bitrate_bps);
config = CreateConfigWithParameters(
{{"maxplaybackrate", "16001"}, {"stereo", "1"}});
EXPECT_EQ(16001, config.max_playback_rate_hz);
EXPECT_EQ(64000, config.bitrate_bps);
}
// Test 24000 < maxplaybackrate triggers Opus full band mode.
TEST(AudioEncoderOpusTest, SetMaxPlaybackRateFb) {
auto config = CreateConfigWithParameters({{"maxplaybackrate", "24001"}});
EXPECT_EQ(24001, config.max_playback_rate_hz);
EXPECT_EQ(32000, config.bitrate_bps);
config = CreateConfigWithParameters(
{{"maxplaybackrate", "24001"}, {"stereo", "1"}});
EXPECT_EQ(24001, config.max_playback_rate_hz);
EXPECT_EQ(64000, config.bitrate_bps);
}
TEST_P(AudioEncoderOpusTest, OpusFlagDtxAsNonSpeech) {
// Create encoder with DTX enabled.
AudioEncoderOpusConfig config;
config.dtx_enabled = true;
config.sample_rate_hz = sample_rate_hz_;
constexpr int payload_type = 17;
const auto encoder = AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
// Open file containing speech and silence.
const std::string kInputFileName =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
test::AudioLoop audio_loop;
// Use the file as if it were sampled at our desired input rate.
const size_t max_loop_length_samples =
sample_rate_hz_ * 10; // Max 10 second loop.
const size_t input_block_size_samples =
10 * sample_rate_hz_ / 1000; // 10 ms.
EXPECT_TRUE(audio_loop.Init(kInputFileName, max_loop_length_samples,
input_block_size_samples));
// Encode.
AudioEncoder::EncodedInfo info;
rtc::Buffer encoded(500);
int nonspeech_frames = 0;
int max_nonspeech_frames = 0;
int dtx_frames = 0;
int max_dtx_frames = 0;
uint32_t rtp_timestamp = 0u;
for (size_t i = 0; i < 500; ++i) {
encoded.Clear();
// Every second call to the encoder will generate an Opus packet.
for (int j = 0; j < 2; j++) {
info =
encoder->Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded);
rtp_timestamp += input_block_size_samples;
}
// Bookkeeping of number of DTX frames.
if (info.encoded_bytes <= 2) {
++dtx_frames;
} else {
if (dtx_frames > max_dtx_frames)
max_dtx_frames = dtx_frames;
dtx_frames = 0;
}
// Bookkeeping of number of non-speech frames.
if (info.speech == 0) {
++nonspeech_frames;
} else {
if (nonspeech_frames > max_nonspeech_frames)
max_nonspeech_frames = nonspeech_frames;
nonspeech_frames = 0;
}
}
// Maximum number of consecutive non-speech packets should exceed 15.
EXPECT_GT(max_nonspeech_frames, 15);
}
TEST(AudioEncoderOpusTest, OpusDtxFilteringHighEnergyRefreshPackets) {
test::ScopedFieldTrials override_field_trials(
"WebRTC-Audio-OpusAvoidNoisePumpingDuringDtx/Enabled/");
const std::string kInputFileName =
webrtc::test::ResourcePath("audio_coding/testfile16kHz", "pcm");
constexpr int kSampleRateHz = 16000;
AudioEncoderOpusConfig config;
config.dtx_enabled = true;
config.sample_rate_hz = kSampleRateHz;
constexpr int payload_type = 17;
const auto encoder = AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
test::AudioLoop audio_loop;
constexpr size_t kMaxLoopLengthSaples = kSampleRateHz * 11.6f;
constexpr size_t kInputBlockSizeSamples = kSampleRateHz / 100;
EXPECT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSaples,
kInputBlockSizeSamples));
AudioEncoder::EncodedInfo info;
rtc::Buffer encoded(500);
// Encode the audio file and store the last part that corresponds to silence.
constexpr size_t kSilenceDurationSamples = kSampleRateHz * 0.2f;
std::array<int16_t, kSilenceDurationSamples> silence;
uint32_t rtp_timestamp = 0;
bool last_packet_dtx_frame = false;
bool opus_entered_dtx = false;
bool silence_filled = false;
size_t timestamp_start_silence = 0;
while (!silence_filled && rtp_timestamp < kMaxLoopLengthSaples) {
encoded.Clear();
// Every second call to the encoder will generate an Opus packet.
for (int j = 0; j < 2; j++) {
auto next_frame = audio_loop.GetNextBlock();
info = encoder->Encode(rtp_timestamp, next_frame, &encoded);
if (opus_entered_dtx) {
size_t silence_frame_start = rtp_timestamp - timestamp_start_silence;
silence_filled = silence_frame_start >= kSilenceDurationSamples;
if (!silence_filled) {
std::copy(next_frame.begin(), next_frame.end(),
silence.begin() + silence_frame_start);
}
}
rtp_timestamp += kInputBlockSizeSamples;
}
EXPECT_TRUE(info.encoded_bytes > 0 || last_packet_dtx_frame);
last_packet_dtx_frame = info.encoded_bytes > 0 ? info.encoded_bytes <= 2
: last_packet_dtx_frame;
if (info.encoded_bytes <= 2 && !opus_entered_dtx) {
timestamp_start_silence = rtp_timestamp;
}
opus_entered_dtx = info.encoded_bytes <= 2;
}
EXPECT_TRUE(silence_filled);
// The copied 200 ms of silence is used for creating 6 bursts that are fed to
// the encoder, the first three ones with a larger energy and the last three
// with a lower energy. This test verifies that the encoder just sends refresh
// DTX packets during the last bursts.
int number_non_empty_packets_during_increase = 0;
int number_non_empty_packets_during_decrease = 0;
for (size_t burst = 0; burst < 6; ++burst) {
uint32_t rtp_timestamp_start = rtp_timestamp;
const bool increase_noise = burst < 3;
const float gain = increase_noise ? 1.4f : 0.0f;
while (rtp_timestamp < rtp_timestamp_start + kSilenceDurationSamples) {
encoded.Clear();
// Every second call to the encoder will generate an Opus packet.
for (int j = 0; j < 2; j++) {
std::array<int16_t, kInputBlockSizeSamples> silence_frame;
size_t silence_frame_start = rtp_timestamp - rtp_timestamp_start;
std::transform(
silence.begin() + silence_frame_start,
silence.begin() + silence_frame_start + kInputBlockSizeSamples,
silence_frame.begin(), [gain](float s) { return gain * s; });
info = encoder->Encode(rtp_timestamp, silence_frame, &encoded);
rtp_timestamp += kInputBlockSizeSamples;
}
EXPECT_TRUE(info.encoded_bytes > 0 || last_packet_dtx_frame);
last_packet_dtx_frame = info.encoded_bytes > 0 ? info.encoded_bytes <= 2
: last_packet_dtx_frame;
// Tracking the number of non empty packets.
if (increase_noise && info.encoded_bytes > 2) {
number_non_empty_packets_during_increase++;
}
if (!increase_noise && info.encoded_bytes > 2) {
number_non_empty_packets_during_decrease++;
}
}
}
// Check that the refresh DTX packets are just sent during the decrease energy
// region.
EXPECT_EQ(number_non_empty_packets_during_increase, 0);
EXPECT_GT(number_non_empty_packets_during_decrease, 0);
}
} // namespace webrtc
Reference in a new issue View git blame Copy permalink