blob: 8955810429fdae57f07de1c8c92d999493c6efc9 [file] [log] [blame]
/*
* Copyright (c) 2020 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/voip/audio_channel.h"
#include "absl/functional/any_invocable.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/call/transport.h"
#include "api/task_queue/task_queue_factory.h"
#include "audio/voip/test/mock_task_queue.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/audio_mixer/sine_wave_generator.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/logging.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/mock_transport.h"
namespace webrtc {
namespace {
using ::testing::Invoke;
using ::testing::NiceMock;
using ::testing::Return;
using ::testing::Unused;
constexpr uint64_t kStartTime = 123456789;
constexpr uint32_t kLocalSsrc = 0xdeadc0de;
constexpr int16_t kAudioLevel = 3004; // used for sine wave level
constexpr int kPcmuPayload = 0;
class AudioChannelTest : public ::testing::Test {
public:
const SdpAudioFormat kPcmuFormat = {"pcmu", 8000, 1};
AudioChannelTest()
: fake_clock_(kStartTime), wave_generator_(1000.0, kAudioLevel) {
task_queue_factory_ = std::make_unique<MockTaskQueueFactory>(&task_queue_);
audio_mixer_ = AudioMixerImpl::Create();
encoder_factory_ = CreateBuiltinAudioEncoderFactory();
decoder_factory_ = CreateBuiltinAudioDecoderFactory();
// By default, run the queued task immediately.
ON_CALL(task_queue_, PostTask)
.WillByDefault(
[](absl::AnyInvocable<void() &&> task) { std::move(task)(); });
}
void SetUp() override { audio_channel_ = CreateAudioChannel(kLocalSsrc); }
void TearDown() override { audio_channel_ = nullptr; }
rtc::scoped_refptr<AudioChannel> CreateAudioChannel(uint32_t ssrc) {
// Use same audio mixer here for simplicity sake as we are not checking
// audio activity of RTP in our testcases. If we need to do test on audio
// signal activity then we need to assign audio mixer for each channel.
// Also this uses the same transport object for different audio channel to
// simplify network routing logic.
rtc::scoped_refptr<AudioChannel> audio_channel =
rtc::make_ref_counted<AudioChannel>(
&transport_, ssrc, task_queue_factory_.get(), audio_mixer_.get(),
decoder_factory_);
audio_channel->SetEncoder(kPcmuPayload, kPcmuFormat,
encoder_factory_->MakeAudioEncoder(
kPcmuPayload, kPcmuFormat, absl::nullopt));
audio_channel->SetReceiveCodecs({{kPcmuPayload, kPcmuFormat}});
audio_channel->StartSend();
audio_channel->StartPlay();
return audio_channel;
}
std::unique_ptr<AudioFrame> GetAudioFrame(int order) {
auto frame = std::make_unique<AudioFrame>();
frame->sample_rate_hz_ = kPcmuFormat.clockrate_hz;
frame->samples_per_channel_ = kPcmuFormat.clockrate_hz / 100; // 10 ms.
frame->num_channels_ = kPcmuFormat.num_channels;
frame->timestamp_ = frame->samples_per_channel_ * order;
wave_generator_.GenerateNextFrame(frame.get());
return frame;
}
SimulatedClock fake_clock_;
SineWaveGenerator wave_generator_;
NiceMock<MockTransport> transport_;
NiceMock<MockTaskQueue> task_queue_;
std::unique_ptr<TaskQueueFactory> task_queue_factory_;
rtc::scoped_refptr<AudioMixer> audio_mixer_;
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
rtc::scoped_refptr<AudioChannel> audio_channel_;
};
// Validate RTP packet generation by feeding audio frames with sine wave.
// Resulted RTP packet is looped back into AudioChannel and gets decoded into
// audio frame to see if it has some signal to indicate its validity.
TEST_F(AudioChannelTest, PlayRtpByLocalLoop) {
auto loop_rtp = [&](const uint8_t* packet, size_t length, Unused) {
audio_channel_->ReceivedRTPPacket(
rtc::ArrayView<const uint8_t>(packet, length));
return true;
};
EXPECT_CALL(transport_, SendRtp).WillOnce(Invoke(loop_rtp));
auto audio_sender = audio_channel_->GetAudioSender();
audio_sender->SendAudioData(GetAudioFrame(0));
audio_sender->SendAudioData(GetAudioFrame(1));
AudioFrame empty_frame, audio_frame;
empty_frame.Mute();
empty_frame.mutable_data(); // This will zero out the data.
audio_frame.CopyFrom(empty_frame);
audio_mixer_->Mix(/*number_of_channels*/ 1, &audio_frame);
// We expect now audio frame to pick up something.
EXPECT_NE(memcmp(empty_frame.data(), audio_frame.data(),
AudioFrame::kMaxDataSizeBytes),
0);
}
// Validate assigned local SSRC is resulted in RTP packet.
TEST_F(AudioChannelTest, VerifyLocalSsrcAsAssigned) {
RtpPacketReceived rtp;
auto loop_rtp = [&](const uint8_t* packet, size_t length, Unused) {
rtp.Parse(packet, length);
return true;
};
EXPECT_CALL(transport_, SendRtp).WillOnce(Invoke(loop_rtp));
auto audio_sender = audio_channel_->GetAudioSender();
audio_sender->SendAudioData(GetAudioFrame(0));
audio_sender->SendAudioData(GetAudioFrame(1));
EXPECT_EQ(rtp.Ssrc(), kLocalSsrc);
}
// Check metrics after processing an RTP packet.
TEST_F(AudioChannelTest, TestIngressStatistics) {
auto loop_rtp = [&](const uint8_t* packet, size_t length, Unused) {
audio_channel_->ReceivedRTPPacket(
rtc::ArrayView<const uint8_t>(packet, length));
return true;
};
EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(loop_rtp));
auto audio_sender = audio_channel_->GetAudioSender();
audio_sender->SendAudioData(GetAudioFrame(0));
audio_sender->SendAudioData(GetAudioFrame(1));
AudioFrame audio_frame;
audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
absl::optional<IngressStatistics> ingress_stats =
audio_channel_->GetIngressStatistics();
EXPECT_TRUE(ingress_stats);
EXPECT_EQ(ingress_stats->neteq_stats.total_samples_received, 160ULL);
EXPECT_EQ(ingress_stats->neteq_stats.concealed_samples, 0ULL);
EXPECT_EQ(ingress_stats->neteq_stats.concealment_events, 0ULL);
EXPECT_EQ(ingress_stats->neteq_stats.inserted_samples_for_deceleration, 0ULL);
EXPECT_EQ(ingress_stats->neteq_stats.removed_samples_for_acceleration, 0ULL);
EXPECT_EQ(ingress_stats->neteq_stats.silent_concealed_samples, 0ULL);
// To extract the jitter buffer length in millisecond, jitter_buffer_delay_ms
// needs to be divided by jitter_buffer_emitted_count (number of samples).
EXPECT_EQ(ingress_stats->neteq_stats.jitter_buffer_delay_ms, 1600ULL);
EXPECT_EQ(ingress_stats->neteq_stats.jitter_buffer_emitted_count, 160ULL);
EXPECT_GT(ingress_stats->neteq_stats.jitter_buffer_target_delay_ms, 0ULL);
EXPECT_EQ(ingress_stats->neteq_stats.interruption_count, 0);
EXPECT_EQ(ingress_stats->neteq_stats.total_interruption_duration_ms, 0);
EXPECT_DOUBLE_EQ(ingress_stats->total_duration, 0.02);
// Now without any RTP pending in jitter buffer pull more.
audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
// Send another RTP packet to intentionally break PLC.
audio_sender->SendAudioData(GetAudioFrame(2));
audio_sender->SendAudioData(GetAudioFrame(3));
ingress_stats = audio_channel_->GetIngressStatistics();
EXPECT_TRUE(ingress_stats);
EXPECT_EQ(ingress_stats->neteq_stats.total_samples_received, 320ULL);
EXPECT_EQ(ingress_stats->neteq_stats.concealed_samples, 168ULL);
EXPECT_EQ(ingress_stats->neteq_stats.concealment_events, 1ULL);
EXPECT_EQ(ingress_stats->neteq_stats.inserted_samples_for_deceleration, 0ULL);
EXPECT_EQ(ingress_stats->neteq_stats.removed_samples_for_acceleration, 0ULL);
EXPECT_EQ(ingress_stats->neteq_stats.silent_concealed_samples, 0ULL);
EXPECT_EQ(ingress_stats->neteq_stats.jitter_buffer_delay_ms, 1600ULL);
EXPECT_EQ(ingress_stats->neteq_stats.jitter_buffer_emitted_count, 160ULL);
EXPECT_GT(ingress_stats->neteq_stats.jitter_buffer_target_delay_ms, 0ULL);
EXPECT_EQ(ingress_stats->neteq_stats.interruption_count, 0);
EXPECT_EQ(ingress_stats->neteq_stats.total_interruption_duration_ms, 0);
EXPECT_DOUBLE_EQ(ingress_stats->total_duration, 0.04);
// Pull the last RTP packet.
audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
ingress_stats = audio_channel_->GetIngressStatistics();
EXPECT_TRUE(ingress_stats);
EXPECT_EQ(ingress_stats->neteq_stats.total_samples_received, 480ULL);
EXPECT_EQ(ingress_stats->neteq_stats.concealed_samples, 168ULL);
EXPECT_EQ(ingress_stats->neteq_stats.concealment_events, 1ULL);
EXPECT_EQ(ingress_stats->neteq_stats.inserted_samples_for_deceleration, 0ULL);
EXPECT_EQ(ingress_stats->neteq_stats.removed_samples_for_acceleration, 0ULL);
EXPECT_EQ(ingress_stats->neteq_stats.silent_concealed_samples, 0ULL);
EXPECT_EQ(ingress_stats->neteq_stats.jitter_buffer_delay_ms, 3200ULL);
EXPECT_EQ(ingress_stats->neteq_stats.jitter_buffer_emitted_count, 320ULL);
EXPECT_GT(ingress_stats->neteq_stats.jitter_buffer_target_delay_ms, 0ULL);
EXPECT_EQ(ingress_stats->neteq_stats.interruption_count, 0);
EXPECT_EQ(ingress_stats->neteq_stats.total_interruption_duration_ms, 0);
EXPECT_DOUBLE_EQ(ingress_stats->total_duration, 0.06);
}
// Check ChannelStatistics metric after processing RTP and RTCP packets.
TEST_F(AudioChannelTest, TestChannelStatistics) {
auto loop_rtp = [&](const uint8_t* packet, size_t length, Unused) {
audio_channel_->ReceivedRTPPacket(
rtc::ArrayView<const uint8_t>(packet, length));
return true;
};
auto loop_rtcp = [&](const uint8_t* packet, size_t length) {
audio_channel_->ReceivedRTCPPacket(
rtc::ArrayView<const uint8_t>(packet, length));
return true;
};
EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(loop_rtp));
EXPECT_CALL(transport_, SendRtcp).WillRepeatedly(Invoke(loop_rtcp));
// Simulate microphone giving audio frame (10 ms). This will trigger tranport
// to send RTP as handled in loop_rtp above.
auto audio_sender = audio_channel_->GetAudioSender();
audio_sender->SendAudioData(GetAudioFrame(0));
audio_sender->SendAudioData(GetAudioFrame(1));
// Simulate speaker requesting audio frame (10 ms). This will trigger VoIP
// engine to fetch audio samples from RTP packets stored in jitter buffer.
AudioFrame audio_frame;
audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
// Force sending RTCP SR report in order to have remote_rtcp field available
// in channel statistics. This will trigger tranport to send RTCP as handled
// in loop_rtcp above.
audio_channel_->SendRTCPReportForTesting(kRtcpSr);
absl::optional<ChannelStatistics> channel_stats =
audio_channel_->GetChannelStatistics();
EXPECT_TRUE(channel_stats);
EXPECT_EQ(channel_stats->packets_sent, 1ULL);
EXPECT_EQ(channel_stats->bytes_sent, 160ULL);
EXPECT_EQ(channel_stats->packets_received, 1ULL);
EXPECT_EQ(channel_stats->bytes_received, 160ULL);
EXPECT_EQ(channel_stats->jitter, 0);
EXPECT_EQ(channel_stats->packets_lost, 0);
EXPECT_EQ(channel_stats->remote_ssrc.value(), kLocalSsrc);
EXPECT_TRUE(channel_stats->remote_rtcp.has_value());
EXPECT_EQ(channel_stats->remote_rtcp->jitter, 0);
EXPECT_EQ(channel_stats->remote_rtcp->packets_lost, 0);
EXPECT_EQ(channel_stats->remote_rtcp->fraction_lost, 0);
EXPECT_GT(channel_stats->remote_rtcp->last_report_received_timestamp_ms, 0);
EXPECT_FALSE(channel_stats->remote_rtcp->round_trip_time.has_value());
}
// Check ChannelStatistics RTT metric after processing RTP and RTCP packets
// using three audio channels where each represents media endpoint.
//
// 1) AC1 <- RTP/RTCP -> AC2
// 2) AC1 <- RTP/RTCP -> AC3
//
// During step 1), AC1 should be able to check RTT from AC2's SSRC.
// During step 2), AC1 should be able to check RTT from AC3's SSRC.
TEST_F(AudioChannelTest, RttIsAvailableAfterChangeOfRemoteSsrc) {
// Create AC2 and AC3.
constexpr uint32_t kAc2Ssrc = 0xdeadbeef;
constexpr uint32_t kAc3Ssrc = 0xdeafbeef;
auto ac_2 = CreateAudioChannel(kAc2Ssrc);
auto ac_3 = CreateAudioChannel(kAc3Ssrc);
auto send_recv_rtp = [&](rtc::scoped_refptr<AudioChannel> rtp_sender,
rtc::scoped_refptr<AudioChannel> rtp_receiver) {
// Setup routing logic via transport_.
auto route_rtp = [&](const uint8_t* packet, size_t length, Unused) {
rtp_receiver->ReceivedRTPPacket(rtc::MakeArrayView(packet, length));
return true;
};
ON_CALL(transport_, SendRtp).WillByDefault(route_rtp);
// This will trigger route_rtp callback via transport_.
rtp_sender->GetAudioSender()->SendAudioData(GetAudioFrame(0));
rtp_sender->GetAudioSender()->SendAudioData(GetAudioFrame(1));
// Process received RTP in receiver.
AudioFrame audio_frame;
audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
// Revert to default to avoid using reference in route_rtp lambda.
ON_CALL(transport_, SendRtp).WillByDefault(Return(true));
};
auto send_recv_rtcp = [&](rtc::scoped_refptr<AudioChannel> rtcp_sender,
rtc::scoped_refptr<AudioChannel> rtcp_receiver) {
// Setup routing logic via transport_.
auto route_rtcp = [&](const uint8_t* packet, size_t length) {
rtcp_receiver->ReceivedRTCPPacket(rtc::MakeArrayView(packet, length));
return true;
};
ON_CALL(transport_, SendRtcp).WillByDefault(route_rtcp);
// This will trigger route_rtcp callback via transport_.
rtcp_sender->SendRTCPReportForTesting(kRtcpSr);
// Revert to default to avoid using reference in route_rtcp lambda.
ON_CALL(transport_, SendRtcp).WillByDefault(Return(true));
};
// AC1 <-- RTP/RTCP --> AC2
send_recv_rtp(audio_channel_, ac_2);
send_recv_rtp(ac_2, audio_channel_);
send_recv_rtcp(audio_channel_, ac_2);
send_recv_rtcp(ac_2, audio_channel_);
absl::optional<ChannelStatistics> channel_stats =
audio_channel_->GetChannelStatistics();
ASSERT_TRUE(channel_stats);
EXPECT_EQ(channel_stats->remote_ssrc, kAc2Ssrc);
ASSERT_TRUE(channel_stats->remote_rtcp);
EXPECT_GT(channel_stats->remote_rtcp->round_trip_time, 0.0);
// AC1 <-- RTP/RTCP --> AC3
send_recv_rtp(audio_channel_, ac_3);
send_recv_rtp(ac_3, audio_channel_);
send_recv_rtcp(audio_channel_, ac_3);
send_recv_rtcp(ac_3, audio_channel_);
channel_stats = audio_channel_->GetChannelStatistics();
ASSERT_TRUE(channel_stats);
EXPECT_EQ(channel_stats->remote_ssrc, kAc3Ssrc);
ASSERT_TRUE(channel_stats->remote_rtcp);
EXPECT_GT(channel_stats->remote_rtcp->round_trip_time, 0.0);
}
} // namespace
} // namespace webrtc