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v811_spc009/external/webrtc/audio/audio_send_stream_unittest.cc

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/*
* 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 "audio/audio_send_stream.h"
#include <memory>
#include <string>
#include <thread>
#include <utility>
#include <vector>
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/mock_frame_encryptor.h"
#include "audio/audio_state.h"
#include "audio/conversion.h"
#include "audio/mock_voe_channel_proxy.h"
#include "call/test/mock_rtp_transport_controller_send.h"
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
#include "modules/audio_device/include/mock_audio_device.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/audio_mixer/sine_wave_generator.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
#include "modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h"
#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
#include "rtc_base/task_queue_for_test.h"
#include "system_wrappers/include/clock.h"
#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/mock_audio_encoder.h"
#include "test/mock_audio_encoder_factory.h"
namespace webrtc {
namespace test {
namespace {
using ::testing::_;
using ::testing::AnyNumber;
using ::testing::Eq;
using ::testing::Field;
using ::testing::Invoke;
using ::testing::Ne;
using ::testing::Return;
using ::testing::StrEq;
static const float kTolerance = 0.0001f;
const uint32_t kSsrc = 1234;
const char* kCName = "foo_name";
const int kAudioLevelId = 2;
const int kTransportSequenceNumberId = 4;
const int32_t kEchoDelayMedian = 254;
const int32_t kEchoDelayStdDev = -3;
const double kDivergentFilterFraction = 0.2f;
const double kEchoReturnLoss = -65;
const double kEchoReturnLossEnhancement = 101;
const double kResidualEchoLikelihood = -1.0f;
const double kResidualEchoLikelihoodMax = 23.0f;
const CallSendStatistics kCallStats = {112, 12, 13456, 17890};
const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
const int kTelephoneEventPayloadType = 123;
const int kTelephoneEventPayloadFrequency = 65432;
const int kTelephoneEventCode = 45;
const int kTelephoneEventDuration = 6789;
constexpr int kIsacPayloadType = 103;
const SdpAudioFormat kIsacFormat = {"isac", 16000, 1};
const SdpAudioFormat kOpusFormat = {"opus", 48000, 2};
const SdpAudioFormat kG722Format = {"g722", 8000, 1};
const AudioCodecSpec kCodecSpecs[] = {
{kIsacFormat, {16000, 1, 32000, 10000, 32000}},
{kOpusFormat, {48000, 1, 32000, 6000, 510000}},
{kG722Format, {16000, 1, 64000}}};
// TODO(dklee): This mirrors calculation in audio_send_stream.cc, which
// should be made more precise in the future. This can be changed when that
// logic is more accurate.
const DataSize kOverheadPerPacket = DataSize::Bytes(20 + 8 + 10 + 12);
const TimeDelta kMinFrameLength = TimeDelta::Millis(20);
const TimeDelta kMaxFrameLength = TimeDelta::Millis(120);
const DataRate kMinOverheadRate = kOverheadPerPacket / kMaxFrameLength;
const DataRate kMaxOverheadRate = kOverheadPerPacket / kMinFrameLength;
class MockLimitObserver : public BitrateAllocator::LimitObserver {
public:
MOCK_METHOD(void,
OnAllocationLimitsChanged,
(BitrateAllocationLimits),
(override));
};
std::unique_ptr<MockAudioEncoder> SetupAudioEncoderMock(
int payload_type,
const SdpAudioFormat& format) {
for (const auto& spec : kCodecSpecs) {
if (format == spec.format) {
std::unique_ptr<MockAudioEncoder> encoder(
new ::testing::NiceMock<MockAudioEncoder>());
ON_CALL(*encoder.get(), SampleRateHz())
.WillByDefault(Return(spec.info.sample_rate_hz));
ON_CALL(*encoder.get(), NumChannels())
.WillByDefault(Return(spec.info.num_channels));
ON_CALL(*encoder.get(), RtpTimestampRateHz())
.WillByDefault(Return(spec.format.clockrate_hz));
ON_CALL(*encoder.get(), GetFrameLengthRange())
.WillByDefault(Return(absl::optional<std::pair<TimeDelta, TimeDelta>>{
{TimeDelta::Millis(20), TimeDelta::Millis(120)}}));
return encoder;
}
}
return nullptr;
}
rtc::scoped_refptr<MockAudioEncoderFactory> SetupEncoderFactoryMock() {
rtc::scoped_refptr<MockAudioEncoderFactory> factory =
new rtc::RefCountedObject<MockAudioEncoderFactory>();
ON_CALL(*factory.get(), GetSupportedEncoders())
.WillByDefault(Return(std::vector<AudioCodecSpec>(
std::begin(kCodecSpecs), std::end(kCodecSpecs))));
ON_CALL(*factory.get(), QueryAudioEncoder(_))
.WillByDefault(Invoke(
[](const SdpAudioFormat& format) -> absl::optional<AudioCodecInfo> {
for (const auto& spec : kCodecSpecs) {
if (format == spec.format) {
return spec.info;
}
}
return absl::nullopt;
}));
ON_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _, _))
.WillByDefault(Invoke([](int payload_type, const SdpAudioFormat& format,
absl::optional<AudioCodecPairId> codec_pair_id,
std::unique_ptr<AudioEncoder>* return_value) {
*return_value = SetupAudioEncoderMock(payload_type, format);
}));
return factory;
}
struct ConfigHelper {
ConfigHelper(bool audio_bwe_enabled,
bool expect_set_encoder_call,
bool use_null_audio_processing)
: clock_(1000000),
task_queue_factory_(CreateDefaultTaskQueueFactory()),
stream_config_(/*send_transport=*/nullptr),
audio_processing_(
use_null_audio_processing
? nullptr
: new rtc::RefCountedObject<MockAudioProcessing>()),
bitrate_allocator_(&limit_observer_),
worker_queue_(task_queue_factory_->CreateTaskQueue(
"ConfigHelper_worker_queue",
TaskQueueFactory::Priority::NORMAL)),
audio_encoder_(nullptr) {
using ::testing::Invoke;
AudioState::Config config;
config.audio_mixer = AudioMixerImpl::Create();
config.audio_processing = audio_processing_;
config.audio_device_module =
new rtc::RefCountedObject<MockAudioDeviceModule>();
audio_state_ = AudioState::Create(config);
SetupDefaultChannelSend(audio_bwe_enabled);
SetupMockForSetupSendCodec(expect_set_encoder_call);
SetupMockForCallEncoder();
// Use ISAC as default codec so as to prevent unnecessary |channel_proxy_|
// calls from the default ctor behavior.
stream_config_.send_codec_spec =
AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat);
stream_config_.rtp.ssrc = kSsrc;
stream_config_.rtp.c_name = kCName;
stream_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
if (audio_bwe_enabled) {
AddBweToConfig(&stream_config_);
}
stream_config_.encoder_factory = SetupEncoderFactoryMock();
stream_config_.min_bitrate_bps = 10000;
stream_config_.max_bitrate_bps = 65000;
}
std::unique_ptr<internal::AudioSendStream> CreateAudioSendStream() {
EXPECT_CALL(rtp_transport_, GetWorkerQueue())
.WillRepeatedly(Return(&worker_queue_));
return std::unique_ptr<internal::AudioSendStream>(
new internal::AudioSendStream(
Clock::GetRealTimeClock(), stream_config_, audio_state_,
task_queue_factory_.get(), &rtp_transport_, &bitrate_allocator_,
&event_log_, absl::nullopt,
std::unique_ptr<voe::ChannelSendInterface>(channel_send_)));
}
AudioSendStream::Config& config() { return stream_config_; }
MockAudioEncoderFactory& mock_encoder_factory() {
return *static_cast<MockAudioEncoderFactory*>(
stream_config_.encoder_factory.get());
}
MockRtpRtcpInterface* rtp_rtcp() { return &rtp_rtcp_; }
MockChannelSend* channel_send() { return channel_send_; }
RtpTransportControllerSendInterface* transport() { return &rtp_transport_; }
static void AddBweToConfig(AudioSendStream::Config* config) {
config->rtp.extensions.push_back(RtpExtension(
RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
config->send_codec_spec->transport_cc_enabled = true;
}
void SetupDefaultChannelSend(bool audio_bwe_enabled) {
EXPECT_TRUE(channel_send_ == nullptr);
channel_send_ = new ::testing::StrictMock<MockChannelSend>();
EXPECT_CALL(*channel_send_, GetRtpRtcp()).WillRepeatedly(Invoke([this]() {
return &this->rtp_rtcp_;
}));
EXPECT_CALL(rtp_rtcp_, SSRC).WillRepeatedly(Return(kSsrc));
EXPECT_CALL(*channel_send_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
EXPECT_CALL(*channel_send_, SetFrameEncryptor(_)).Times(1);
EXPECT_CALL(*channel_send_, SetEncoderToPacketizerFrameTransformer(_))
.Times(1);
EXPECT_CALL(rtp_rtcp_, SetExtmapAllowMixed(false)).Times(1);
EXPECT_CALL(*channel_send_,
SetSendAudioLevelIndicationStatus(true, kAudioLevelId))
.Times(1);
EXPECT_CALL(rtp_transport_, GetBandwidthObserver())
.WillRepeatedly(Return(&bandwidth_observer_));
if (audio_bwe_enabled) {
EXPECT_CALL(rtp_rtcp_,
RegisterRtpHeaderExtension(TransportSequenceNumber::kUri,
kTransportSequenceNumberId))
.Times(1);
EXPECT_CALL(*channel_send_,
RegisterSenderCongestionControlObjects(
&rtp_transport_, Eq(&bandwidth_observer_)))
.Times(1);
} else {
EXPECT_CALL(*channel_send_, RegisterSenderCongestionControlObjects(
&rtp_transport_, Eq(nullptr)))
.Times(1);
}
EXPECT_CALL(*channel_send_, ResetSenderCongestionControlObjects()).Times(1);
EXPECT_CALL(rtp_rtcp_, SetRid(std::string())).Times(1);
}
void SetupMockForSetupSendCodec(bool expect_set_encoder_call) {
if (expect_set_encoder_call) {
EXPECT_CALL(*channel_send_, SetEncoder)
.WillOnce(
[this](int payload_type, std::unique_ptr<AudioEncoder> encoder) {
this->audio_encoder_ = std::move(encoder);
return true;
});
}
}
void SetupMockForCallEncoder() {
// Let ModifyEncoder to invoke mock audio encoder.
EXPECT_CALL(*channel_send_, CallEncoder(_))
.WillRepeatedly(
[this](rtc::FunctionView<void(AudioEncoder*)> modifier) {
if (this->audio_encoder_)
modifier(this->audio_encoder_.get());
});
}
void SetupMockForSendTelephoneEvent() {
EXPECT_TRUE(channel_send_);
EXPECT_CALL(*channel_send_, SetSendTelephoneEventPayloadType(
kTelephoneEventPayloadType,
kTelephoneEventPayloadFrequency));
EXPECT_CALL(
*channel_send_,
SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration))
.WillOnce(Return(true));
}
void SetupMockForGetStats(bool use_null_audio_processing) {
using ::testing::DoAll;
using ::testing::SetArgPointee;
using ::testing::SetArgReferee;
std::vector<ReportBlock> report_blocks;
webrtc::ReportBlock block = kReportBlock;
report_blocks.push_back(block); // Has wrong SSRC.
block.source_SSRC = kSsrc;
report_blocks.push_back(block); // Correct block.
block.fraction_lost = 0;
report_blocks.push_back(block); // Duplicate SSRC, bad fraction_lost.
EXPECT_TRUE(channel_send_);
EXPECT_CALL(*channel_send_, GetRTCPStatistics())
.WillRepeatedly(Return(kCallStats));
EXPECT_CALL(*channel_send_, GetRemoteRTCPReportBlocks())
.WillRepeatedly(Return(report_blocks));
EXPECT_CALL(*channel_send_, GetANAStatistics())
.WillRepeatedly(Return(ANAStats()));
EXPECT_CALL(*channel_send_, GetBitrate()).WillRepeatedly(Return(0));
audio_processing_stats_.echo_return_loss = kEchoReturnLoss;
audio_processing_stats_.echo_return_loss_enhancement =
kEchoReturnLossEnhancement;
audio_processing_stats_.delay_median_ms = kEchoDelayMedian;
audio_processing_stats_.delay_standard_deviation_ms = kEchoDelayStdDev;
audio_processing_stats_.divergent_filter_fraction =
kDivergentFilterFraction;
audio_processing_stats_.residual_echo_likelihood = kResidualEchoLikelihood;
audio_processing_stats_.residual_echo_likelihood_recent_max =
kResidualEchoLikelihoodMax;
if (!use_null_audio_processing) {
ASSERT_TRUE(audio_processing_);
EXPECT_CALL(*audio_processing_, GetStatistics(true))
.WillRepeatedly(Return(audio_processing_stats_));
}
}
TaskQueueForTest* worker() { return &worker_queue_; }
private:
SimulatedClock clock_;
std::unique_ptr<TaskQueueFactory> task_queue_factory_;
rtc::scoped_refptr<AudioState> audio_state_;
AudioSendStream::Config stream_config_;
::testing::StrictMock<MockChannelSend>* channel_send_ = nullptr;
rtc::scoped_refptr<MockAudioProcessing> audio_processing_;
AudioProcessingStats audio_processing_stats_;
::testing::StrictMock<MockRtcpBandwidthObserver> bandwidth_observer_;
::testing::NiceMock<MockRtcEventLog> event_log_;
::testing::NiceMock<MockRtpTransportControllerSend> rtp_transport_;
::testing::NiceMock<MockRtpRtcpInterface> rtp_rtcp_;
::testing::NiceMock<MockLimitObserver> limit_observer_;
BitrateAllocator bitrate_allocator_;
// |worker_queue| is defined last to ensure all pending tasks are cancelled
// and deleted before any other members.
TaskQueueForTest worker_queue_;
std::unique_ptr<AudioEncoder> audio_encoder_;
};
// The audio level ranges linearly [0,32767].
std::unique_ptr<AudioFrame> CreateAudioFrame1kHzSineWave(int16_t audio_level,
int duration_ms,
int sample_rate_hz,
size_t num_channels) {
size_t samples_per_channel = sample_rate_hz / (1000 / duration_ms);
std::vector<int16_t> audio_data(samples_per_channel * num_channels, 0);
std::unique_ptr<AudioFrame> audio_frame = std::make_unique<AudioFrame>();
audio_frame->UpdateFrame(0 /* RTP timestamp */, &audio_data[0],
samples_per_channel, sample_rate_hz,
AudioFrame::SpeechType::kNormalSpeech,
AudioFrame::VADActivity::kVadUnknown, num_channels);
SineWaveGenerator wave_generator(1000.0, audio_level);
wave_generator.GenerateNextFrame(audio_frame.get());
return audio_frame;
}
} // namespace
TEST(AudioSendStreamTest, ConfigToString) {
AudioSendStream::Config config(/*send_transport=*/nullptr);
config.rtp.ssrc = kSsrc;
config.rtp.c_name = kCName;
config.min_bitrate_bps = 12000;
config.max_bitrate_bps = 34000;
config.send_codec_spec =
AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat);
config.send_codec_spec->nack_enabled = true;
config.send_codec_spec->transport_cc_enabled = false;
config.send_codec_spec->cng_payload_type = 42;
config.send_codec_spec->red_payload_type = 43;
config.encoder_factory = MockAudioEncoderFactory::CreateUnusedFactory();
config.rtp.extmap_allow_mixed = true;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
config.rtcp_report_interval_ms = 2500;
EXPECT_EQ(
"{rtp: {ssrc: 1234, extmap-allow-mixed: true, extensions: [{uri: "
"urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], "
"c_name: foo_name}, rtcp_report_interval_ms: 2500, "
"send_transport: null, "
"min_bitrate_bps: 12000, max_bitrate_bps: 34000, "
"send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
"cng_payload_type: 42, red_payload_type: 43, payload_type: 103, "
"format: {name: isac, clockrate_hz: 16000, num_channels: 1, "
"parameters: {}}}}",
config.ToString());
}
TEST(AudioSendStreamTest, ConstructDestruct) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(false, true, use_null_audio_processing);
auto send_stream = helper.CreateAudioSendStream();
}
}
TEST(AudioSendStreamTest, SendTelephoneEvent) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(false, true, use_null_audio_processing);
auto send_stream = helper.CreateAudioSendStream();
helper.SetupMockForSendTelephoneEvent();
EXPECT_TRUE(send_stream->SendTelephoneEvent(
kTelephoneEventPayloadType, kTelephoneEventPayloadFrequency,
kTelephoneEventCode, kTelephoneEventDuration));
}
}
TEST(AudioSendStreamTest, SetMuted) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(false, true, use_null_audio_processing);
auto send_stream = helper.CreateAudioSendStream();
EXPECT_CALL(*helper.channel_send(), SetInputMute(true));
send_stream->SetMuted(true);
}
}
TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) {
ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/");
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(true, true, use_null_audio_processing);
auto send_stream = helper.CreateAudioSendStream();
}
}
TEST(AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(false, true, use_null_audio_processing);
auto send_stream = helper.CreateAudioSendStream();
}
}
TEST(AudioSendStreamTest, GetStats) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(false, true, use_null_audio_processing);
auto send_stream = helper.CreateAudioSendStream();
helper.SetupMockForGetStats(use_null_audio_processing);
AudioSendStream::Stats stats = send_stream->GetStats(true);
EXPECT_EQ(kSsrc, stats.local_ssrc);
EXPECT_EQ(kCallStats.payload_bytes_sent, stats.payload_bytes_sent);
EXPECT_EQ(kCallStats.header_and_padding_bytes_sent,
stats.header_and_padding_bytes_sent);
EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent);
EXPECT_EQ(kReportBlock.cumulative_num_packets_lost, stats.packets_lost);
EXPECT_EQ(Q8ToFloat(kReportBlock.fraction_lost), stats.fraction_lost);
EXPECT_EQ(kIsacFormat.name, stats.codec_name);
EXPECT_EQ(static_cast<int32_t>(kReportBlock.interarrival_jitter /
(kIsacFormat.clockrate_hz / 1000)),
stats.jitter_ms);
EXPECT_EQ(kCallStats.rttMs, stats.rtt_ms);
EXPECT_EQ(0, stats.audio_level);
EXPECT_EQ(0, stats.total_input_energy);
EXPECT_EQ(0, stats.total_input_duration);
if (!use_null_audio_processing) {
EXPECT_EQ(kEchoDelayMedian, stats.apm_statistics.delay_median_ms);
EXPECT_EQ(kEchoDelayStdDev,
stats.apm_statistics.delay_standard_deviation_ms);
EXPECT_EQ(kEchoReturnLoss, stats.apm_statistics.echo_return_loss);
EXPECT_EQ(kEchoReturnLossEnhancement,
stats.apm_statistics.echo_return_loss_enhancement);
EXPECT_EQ(kDivergentFilterFraction,
stats.apm_statistics.divergent_filter_fraction);
EXPECT_EQ(kResidualEchoLikelihood,
stats.apm_statistics.residual_echo_likelihood);
EXPECT_EQ(kResidualEchoLikelihoodMax,
stats.apm_statistics.residual_echo_likelihood_recent_max);
EXPECT_FALSE(stats.typing_noise_detected);
}
}
}
TEST(AudioSendStreamTest, GetStatsAudioLevel) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(false, true, use_null_audio_processing);
auto send_stream = helper.CreateAudioSendStream();
helper.SetupMockForGetStats(use_null_audio_processing);
EXPECT_CALL(*helper.channel_send(), ProcessAndEncodeAudio)
.Times(AnyNumber());
constexpr int kSampleRateHz = 48000;
constexpr size_t kNumChannels = 1;
constexpr int16_t kSilentAudioLevel = 0;
constexpr int16_t kMaxAudioLevel = 32767; // Audio level is [0,32767].
constexpr int kAudioFrameDurationMs = 10;
// Process 10 audio frames (100 ms) of silence. After this, on the next
// (11-th) frame, the audio level will be updated with the maximum audio
// level of the first 11 frames. See AudioLevel.
for (size_t i = 0; i < 10; ++i) {
send_stream->SendAudioData(
CreateAudioFrame1kHzSineWave(kSilentAudioLevel, kAudioFrameDurationMs,
kSampleRateHz, kNumChannels));
}
AudioSendStream::Stats stats = send_stream->GetStats();
EXPECT_EQ(kSilentAudioLevel, stats.audio_level);
EXPECT_NEAR(0.0f, stats.total_input_energy, kTolerance);
EXPECT_NEAR(0.1f, stats.total_input_duration,
kTolerance); // 100 ms = 0.1 s
// Process 10 audio frames (100 ms) of maximum audio level.
// Note that AudioLevel updates the audio level every 11th frame, processing
// 10 frames above was needed to see a non-zero audio level here.
for (size_t i = 0; i < 10; ++i) {
send_stream->SendAudioData(CreateAudioFrame1kHzSineWave(
kMaxAudioLevel, kAudioFrameDurationMs, kSampleRateHz, kNumChannels));
}
stats = send_stream->GetStats();
EXPECT_EQ(kMaxAudioLevel, stats.audio_level);
// Energy increases by energy*duration, where energy is audio level in
// [0,1].
EXPECT_NEAR(0.1f, stats.total_input_energy, kTolerance); // 0.1 s of max
EXPECT_NEAR(0.2f, stats.total_input_duration,
kTolerance); // 200 ms = 0.2 s
}
}
TEST(AudioSendStreamTest, SendCodecAppliesAudioNetworkAdaptor) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(false, true, use_null_audio_processing);
helper.config().send_codec_spec =
AudioSendStream::Config::SendCodecSpec(0, kOpusFormat);
const std::string kAnaConfigString = "abcde";
const std::string kAnaReconfigString = "12345";
helper.config().rtp.extensions.push_back(RtpExtension(
RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
helper.config().audio_network_adaptor_config = kAnaConfigString;
EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _, _))
.WillOnce(Invoke([&kAnaConfigString, &kAnaReconfigString](
int payload_type, const SdpAudioFormat& format,
absl::optional<AudioCodecPairId> codec_pair_id,
std::unique_ptr<AudioEncoder>* return_value) {
auto mock_encoder = SetupAudioEncoderMock(payload_type, format);
EXPECT_CALL(*mock_encoder,
EnableAudioNetworkAdaptor(StrEq(kAnaConfigString), _))
.WillOnce(Return(true));
EXPECT_CALL(*mock_encoder,
EnableAudioNetworkAdaptor(StrEq(kAnaReconfigString), _))
.WillOnce(Return(true));
*return_value = std::move(mock_encoder);
}));
auto send_stream = helper.CreateAudioSendStream();
auto stream_config = helper.config();
stream_config.audio_network_adaptor_config = kAnaReconfigString;
send_stream->Reconfigure(stream_config);
}
}
// VAD is applied when codec is mono and the CNG frequency matches the codec
// clock rate.
TEST(AudioSendStreamTest, SendCodecCanApplyVad) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(false, false, use_null_audio_processing);
helper.config().send_codec_spec =
AudioSendStream::Config::SendCodecSpec(9, kG722Format);
helper.config().send_codec_spec->cng_payload_type = 105;
std::unique_ptr<AudioEncoder> stolen_encoder;
EXPECT_CALL(*helper.channel_send(), SetEncoder)
.WillOnce([&stolen_encoder](int payload_type,
std::unique_ptr<AudioEncoder> encoder) {
stolen_encoder = std::move(encoder);
return true;
});
EXPECT_CALL(*helper.channel_send(), RegisterCngPayloadType(105, 8000));
auto send_stream = helper.CreateAudioSendStream();
// We cannot truly determine if the encoder created is an AudioEncoderCng.
// It is the only reasonable implementation that will return something from
// ReclaimContainedEncoders, though.
ASSERT_TRUE(stolen_encoder);
EXPECT_FALSE(stolen_encoder->ReclaimContainedEncoders().empty());
}
}
TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(false, true, use_null_audio_processing);
auto send_stream = helper.CreateAudioSendStream();
EXPECT_CALL(
*helper.channel_send(),
OnBitrateAllocation(
Field(&BitrateAllocationUpdate::target_bitrate,
Eq(DataRate::BitsPerSec(helper.config().max_bitrate_bps)))));
BitrateAllocationUpdate update;
update.target_bitrate =
DataRate::BitsPerSec(helper.config().max_bitrate_bps + 5000);
update.packet_loss_ratio = 0;
update.round_trip_time = TimeDelta::Millis(50);
update.bwe_period = TimeDelta::Millis(6000);
helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); },
RTC_FROM_HERE);
}
}
TEST(AudioSendStreamTest, SSBweTargetInRangeRespected) {
ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/");
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(true, true, use_null_audio_processing);
auto send_stream = helper.CreateAudioSendStream();
EXPECT_CALL(
*helper.channel_send(),
OnBitrateAllocation(Field(
&BitrateAllocationUpdate::target_bitrate,
Eq(DataRate::BitsPerSec(helper.config().max_bitrate_bps - 5000)))));
BitrateAllocationUpdate update;
update.target_bitrate =
DataRate::BitsPerSec(helper.config().max_bitrate_bps - 5000);
helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); },
RTC_FROM_HERE);
}
}
TEST(AudioSendStreamTest, SSBweFieldTrialMinRespected) {
ScopedFieldTrials field_trials(
"WebRTC-Audio-SendSideBwe/Enabled/"
"WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(true, true, use_null_audio_processing);
auto send_stream = helper.CreateAudioSendStream();
EXPECT_CALL(
*helper.channel_send(),
OnBitrateAllocation(Field(&BitrateAllocationUpdate::target_bitrate,
Eq(DataRate::KilobitsPerSec(6)))));
BitrateAllocationUpdate update;
update.target_bitrate = DataRate::KilobitsPerSec(1);
helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); },
RTC_FROM_HERE);
}
}
TEST(AudioSendStreamTest, SSBweFieldTrialMaxRespected) {
ScopedFieldTrials field_trials(
"WebRTC-Audio-SendSideBwe/Enabled/"
"WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(true, true, use_null_audio_processing);
auto send_stream = helper.CreateAudioSendStream();
EXPECT_CALL(
*helper.channel_send(),
OnBitrateAllocation(Field(&BitrateAllocationUpdate::target_bitrate,
Eq(DataRate::KilobitsPerSec(64)))));
BitrateAllocationUpdate update;
update.target_bitrate = DataRate::KilobitsPerSec(128);
helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); },
RTC_FROM_HERE);
}
}
TEST(AudioSendStreamTest, SSBweWithOverhead) {
ScopedFieldTrials field_trials(
"WebRTC-Audio-SendSideBwe/Enabled/"
"WebRTC-SendSideBwe-WithOverhead/Enabled/"
"WebRTC-Audio-LegacyOverhead/Disabled/");
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(true, true, use_null_audio_processing);
EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
.WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>()));
auto send_stream = helper.CreateAudioSendStream();
const DataRate bitrate =
DataRate::BitsPerSec(helper.config().max_bitrate_bps) +
kMaxOverheadRate;
EXPECT_CALL(*helper.channel_send(),
OnBitrateAllocation(Field(
&BitrateAllocationUpdate::target_bitrate, Eq(bitrate))));
BitrateAllocationUpdate update;
update.target_bitrate = bitrate;
helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); },
RTC_FROM_HERE);
}
}
TEST(AudioSendStreamTest, SSBweWithOverheadMinRespected) {
ScopedFieldTrials field_trials(
"WebRTC-Audio-SendSideBwe/Enabled/"
"WebRTC-SendSideBwe-WithOverhead/Enabled/"
"WebRTC-Audio-LegacyOverhead/Disabled/"
"WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(true, true, use_null_audio_processing);
EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
.WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>()));
auto send_stream = helper.CreateAudioSendStream();
const DataRate bitrate = DataRate::KilobitsPerSec(6) + kMinOverheadRate;
EXPECT_CALL(*helper.channel_send(),
OnBitrateAllocation(Field(
&BitrateAllocationUpdate::target_bitrate, Eq(bitrate))));
BitrateAllocationUpdate update;
update.target_bitrate = DataRate::KilobitsPerSec(1);
helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); },
RTC_FROM_HERE);
}
}
TEST(AudioSendStreamTest, SSBweWithOverheadMaxRespected) {
ScopedFieldTrials field_trials(
"WebRTC-Audio-SendSideBwe/Enabled/"
"WebRTC-SendSideBwe-WithOverhead/Enabled/"
"WebRTC-Audio-LegacyOverhead/Disabled/"
"WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(true, true, use_null_audio_processing);
EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
.WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>()));
auto send_stream = helper.CreateAudioSendStream();
const DataRate bitrate = DataRate::KilobitsPerSec(64) + kMaxOverheadRate;
EXPECT_CALL(*helper.channel_send(),
OnBitrateAllocation(Field(
&BitrateAllocationUpdate::target_bitrate, Eq(bitrate))));
BitrateAllocationUpdate update;
update.target_bitrate = DataRate::KilobitsPerSec(128);
helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); },
RTC_FROM_HERE);
}
}
TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(false, true, use_null_audio_processing);
auto send_stream = helper.CreateAudioSendStream();
EXPECT_CALL(*helper.channel_send(),
OnBitrateAllocation(Field(&BitrateAllocationUpdate::bwe_period,
Eq(TimeDelta::Millis(5000)))));
BitrateAllocationUpdate update;
update.target_bitrate =
DataRate::BitsPerSec(helper.config().max_bitrate_bps + 5000);
update.packet_loss_ratio = 0;
update.round_trip_time = TimeDelta::Millis(50);
update.bwe_period = TimeDelta::Millis(5000);
helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); },
RTC_FROM_HERE);
}
}
// Test that AudioSendStream doesn't recreate the encoder unnecessarily.
TEST(AudioSendStreamTest, DontRecreateEncoder) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(false, false, use_null_audio_processing);
// WillOnce is (currently) the default used by ConfigHelper if asked to set
// an expectation for SetEncoder. Since this behavior is essential for this
// test to be correct, it's instead set-up manually here. Otherwise a simple
// change to ConfigHelper (say to WillRepeatedly) would silently make this
// test useless.
EXPECT_CALL(*helper.channel_send(), SetEncoder).WillOnce(Return());
EXPECT_CALL(*helper.channel_send(), RegisterCngPayloadType(105, 8000));
helper.config().send_codec_spec =
AudioSendStream::Config::SendCodecSpec(9, kG722Format);
helper.config().send_codec_spec->cng_payload_type = 105;
auto send_stream = helper.CreateAudioSendStream();
send_stream->Reconfigure(helper.config());
}
}
TEST(AudioSendStreamTest, ReconfigureTransportCcResetsFirst) {
ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/");
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(false, true, use_null_audio_processing);
auto send_stream = helper.CreateAudioSendStream();
auto new_config = helper.config();
ConfigHelper::AddBweToConfig(&new_config);
EXPECT_CALL(*helper.rtp_rtcp(),
RegisterRtpHeaderExtension(TransportSequenceNumber::kUri,
kTransportSequenceNumberId))
.Times(1);
{
::testing::InSequence seq;
EXPECT_CALL(*helper.channel_send(), ResetSenderCongestionControlObjects())
.Times(1);
EXPECT_CALL(*helper.channel_send(),
RegisterSenderCongestionControlObjects(helper.transport(),
Ne(nullptr)))
.Times(1);
}
send_stream->Reconfigure(new_config);
}
}
TEST(AudioSendStreamTest, OnTransportOverheadChanged) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(false, true, use_null_audio_processing);
auto send_stream = helper.CreateAudioSendStream();
auto new_config = helper.config();
// CallEncoder will be called on overhead change.
EXPECT_CALL(*helper.channel_send(), CallEncoder);
const size_t transport_overhead_per_packet_bytes = 333;
send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes);
EXPECT_EQ(transport_overhead_per_packet_bytes,
send_stream->TestOnlyGetPerPacketOverheadBytes());
}
}
TEST(AudioSendStreamTest, DoesntCallEncoderWhenOverheadUnchanged) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(false, true, use_null_audio_processing);
auto send_stream = helper.CreateAudioSendStream();
auto new_config = helper.config();
// CallEncoder will be called on overhead change.
EXPECT_CALL(*helper.channel_send(), CallEncoder);
const size_t transport_overhead_per_packet_bytes = 333;
send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes);
// Set the same overhead again, CallEncoder should not be called again.
EXPECT_CALL(*helper.channel_send(), CallEncoder).Times(0);
send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes);
// New overhead, call CallEncoder again
EXPECT_CALL(*helper.channel_send(), CallEncoder);
send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes + 1);
}
}
TEST(AudioSendStreamTest, AudioOverheadChanged) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(false, true, use_null_audio_processing);
const size_t audio_overhead_per_packet_bytes = 555;
EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
.WillRepeatedly(Return(audio_overhead_per_packet_bytes));
auto send_stream = helper.CreateAudioSendStream();
auto new_config = helper.config();
BitrateAllocationUpdate update;
update.target_bitrate =
DataRate::BitsPerSec(helper.config().max_bitrate_bps) +
kMaxOverheadRate;
EXPECT_CALL(*helper.channel_send(), OnBitrateAllocation);
helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); },
RTC_FROM_HERE);
EXPECT_EQ(audio_overhead_per_packet_bytes,
send_stream->TestOnlyGetPerPacketOverheadBytes());
EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
.WillRepeatedly(Return(audio_overhead_per_packet_bytes + 20));
EXPECT_CALL(*helper.channel_send(), OnBitrateAllocation);
helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); },
RTC_FROM_HERE);
EXPECT_EQ(audio_overhead_per_packet_bytes + 20,
send_stream->TestOnlyGetPerPacketOverheadBytes());
}
}
TEST(AudioSendStreamTest, OnAudioAndTransportOverheadChanged) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(false, true, use_null_audio_processing);
const size_t audio_overhead_per_packet_bytes = 555;
EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
.WillRepeatedly(Return(audio_overhead_per_packet_bytes));
auto send_stream = helper.CreateAudioSendStream();
auto new_config = helper.config();
const size_t transport_overhead_per_packet_bytes = 333;
send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes);
BitrateAllocationUpdate update;
update.target_bitrate =
DataRate::BitsPerSec(helper.config().max_bitrate_bps) +
kMaxOverheadRate;
EXPECT_CALL(*helper.channel_send(), OnBitrateAllocation);
helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); },
RTC_FROM_HERE);
EXPECT_EQ(
transport_overhead_per_packet_bytes + audio_overhead_per_packet_bytes,
send_stream->TestOnlyGetPerPacketOverheadBytes());
}
}
// Validates that reconfiguring the AudioSendStream with a Frame encryptor
// correctly reconfigures on the object without crashing.
TEST(AudioSendStreamTest, ReconfigureWithFrameEncryptor) {
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(false, true, use_null_audio_processing);
auto send_stream = helper.CreateAudioSendStream();
auto new_config = helper.config();
rtc::scoped_refptr<FrameEncryptorInterface> mock_frame_encryptor_0(
new rtc::RefCountedObject<MockFrameEncryptor>());
new_config.frame_encryptor = mock_frame_encryptor_0;
EXPECT_CALL(*helper.channel_send(), SetFrameEncryptor(Ne(nullptr)))
.Times(1);
send_stream->Reconfigure(new_config);
// Not updating the frame encryptor shouldn't force it to reconfigure.
EXPECT_CALL(*helper.channel_send(), SetFrameEncryptor(_)).Times(0);
send_stream->Reconfigure(new_config);
// Updating frame encryptor to a new object should force a call to the
// proxy.
rtc::scoped_refptr<FrameEncryptorInterface> mock_frame_encryptor_1(
new rtc::RefCountedObject<MockFrameEncryptor>());
new_config.frame_encryptor = mock_frame_encryptor_1;
new_config.crypto_options.sframe.require_frame_encryption = true;
EXPECT_CALL(*helper.channel_send(), SetFrameEncryptor(Ne(nullptr)))
.Times(1);
send_stream->Reconfigure(new_config);
}
}
} // namespace test
} // namespace webrtc
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