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diff --git a/audio/voip/test/audio_egress_unittest.cc b/audio/voip/test/audio_egress_unittest.cc
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+//
+// 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_egress.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/call/transport.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "modules/audio_mixer/sine_wave_generator.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/event.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::Unused;
+
+std::unique_ptr<RtpRtcp> CreateRtpStack(Clock* clock,
+ Transport* transport,
+ uint32_t remote_ssrc) {
+ RtpRtcp::Configuration rtp_config;
+ rtp_config.clock = clock;
+ rtp_config.audio = true;
+ rtp_config.rtcp_report_interval_ms = 5000;
+ rtp_config.outgoing_transport = transport;
+ rtp_config.local_media_ssrc = remote_ssrc;
+ auto rtp_rtcp = RtpRtcp::Create(rtp_config);
+ rtp_rtcp->SetSendingMediaStatus(false);
+ rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
+ return rtp_rtcp;
+}
+
+// AudioEgressTest configures audio egress by using Rtp Stack, fake clock,
+// and task queue factory. Encoder factory is needed to create codec and
+// configure the RTP stack in audio egress.
+class AudioEgressTest : public ::testing::Test {
+ public:
+ static constexpr int16_t kAudioLevel = 3004; // Used for sine wave level.
+ static constexpr uint16_t kSeqNum = 12345;
+ static constexpr uint64_t kStartTime = 123456789;
+ static constexpr uint32_t kRemoteSsrc = 0xDEADBEEF;
+ const SdpAudioFormat kPcmuFormat = {"pcmu", 8000, 1};
+
+ AudioEgressTest()
+ : fake_clock_(kStartTime), wave_generator_(1000.0, kAudioLevel) {
+ rtp_rtcp_ = CreateRtpStack(&fake_clock_, &transport_, kRemoteSsrc);
+ task_queue_factory_ = CreateDefaultTaskQueueFactory();
+ encoder_factory_ = CreateBuiltinAudioEncoderFactory();
+ }
+
+ // Prepare test on audio egress by using PCMu codec with specific
+ // sequence number and its status to be running.
+ void SetUp() override {
+ egress_ = std::make_unique<AudioEgress>(rtp_rtcp_.get(), &fake_clock_,
+ task_queue_factory_.get());
+ constexpr int kPcmuPayload = 0;
+ egress_->SetEncoder(kPcmuPayload, kPcmuFormat,
+ encoder_factory_->MakeAudioEncoder(
+ kPcmuPayload, kPcmuFormat, absl::nullopt));
+ egress_->StartSend();
+ rtp_rtcp_->SetSequenceNumber(kSeqNum);
+ rtp_rtcp_->SetSendingStatus(true);
+ }
+
+ // Make sure we have shut down rtp stack and reset egress for each test.
+ void TearDown() override {
+ rtp_rtcp_->SetSendingStatus(false);
+ egress_.reset();
+ }
+
+ // Create an audio frame prepared for pcmu encoding. Timestamp is
+ // increased per RTP specification which is the number of samples it contains.
+ // Wave generator writes sine wave which has expected high level set
+ // by kAudioLevel.
+ 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 doesn't directly affect this testcase as the the
+ // AudioFrame's timestamp is driven by GetAudioFrame.
+ SimulatedClock fake_clock_;
+ NiceMock<MockTransport> transport_;
+ SineWaveGenerator wave_generator_;
+ std::unique_ptr<AudioEgress> egress_;
+ std::unique_ptr<TaskQueueFactory> task_queue_factory_;
+ std::unique_ptr<RtpRtcp> rtp_rtcp_;
+ rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
+};
+
+TEST_F(AudioEgressTest, SendingStatusAfterStartAndStop) {
+ EXPECT_TRUE(egress_->IsSending());
+ egress_->StopSend();
+ EXPECT_FALSE(egress_->IsSending());
+}
+
+TEST_F(AudioEgressTest, ProcessAudioWithMute) {
+ constexpr int kExpected = 10;
+ rtc::Event event;
+ int rtp_count = 0;
+ RtpPacketReceived rtp;
+ auto rtp_sent = [&](const uint8_t* packet, size_t length, Unused) {
+ rtp.Parse(packet, length);
+ if (++rtp_count == kExpected) {
+ event.Set();
+ }
+ return true;
+ };
+
+ EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(rtp_sent));
+
+ egress_->SetMute(true);
+
+ // Two 10 ms audio frames will result in rtp packet with ptime 20.
+ for (size_t i = 0; i < kExpected * 2; i++) {
+ egress_->SendAudioData(GetAudioFrame(i));
+ fake_clock_.AdvanceTimeMilliseconds(10);
+ }
+
+ event.Wait(/*ms=*/1000);
+ EXPECT_EQ(rtp_count, kExpected);
+
+ // we expect on pcmu payload to result in 255 for silenced payload
+ RTPHeader header;
+ rtp.GetHeader(&header);
+ size_t packet_length = rtp.size();
+ size_t payload_length = packet_length - header.headerLength;
+ size_t payload_data_length = payload_length - header.paddingLength;
+ const uint8_t* payload = rtp.data() + header.headerLength;
+ for (size_t i = 0; i < payload_data_length; ++i) {
+ EXPECT_EQ(*payload++, 255);
+ }
+}
+
+TEST_F(AudioEgressTest, ProcessAudioWithSineWave) {
+ constexpr int kExpected = 10;
+ rtc::Event event;
+ int rtp_count = 0;
+ RtpPacketReceived rtp;
+ auto rtp_sent = [&](const uint8_t* packet, size_t length, Unused) {
+ rtp.Parse(packet, length);
+ if (++rtp_count == kExpected) {
+ event.Set();
+ }
+ return true;
+ };
+
+ EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(rtp_sent));
+
+ // Two 10 ms audio frames will result in rtp packet with ptime 20.
+ for (size_t i = 0; i < kExpected * 2; i++) {
+ egress_->SendAudioData(GetAudioFrame(i));
+ fake_clock_.AdvanceTimeMilliseconds(10);
+ }
+
+ event.Wait(/*ms=*/1000);
+ EXPECT_EQ(rtp_count, kExpected);
+
+ // we expect on pcmu to result in < 255 for payload with sine wave
+ RTPHeader header;
+ rtp.GetHeader(&header);
+ size_t packet_length = rtp.size();
+ size_t payload_length = packet_length - header.headerLength;
+ size_t payload_data_length = payload_length - header.paddingLength;
+ const uint8_t* payload = rtp.data() + header.headerLength;
+ for (size_t i = 0; i < payload_data_length; ++i) {
+ EXPECT_NE(*payload++, 255);
+ }
+}
+
+TEST_F(AudioEgressTest, SkipAudioEncodingAfterStopSend) {
+ constexpr int kExpected = 10;
+ rtc::Event event;
+ int rtp_count = 0;
+ auto rtp_sent = [&](const uint8_t* packet, size_t length, Unused) {
+ if (++rtp_count == kExpected) {
+ event.Set();
+ }
+ return true;
+ };
+
+ EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(rtp_sent));
+
+ // Two 10 ms audio frames will result in rtp packet with ptime 20.
+ for (size_t i = 0; i < kExpected * 2; i++) {
+ egress_->SendAudioData(GetAudioFrame(i));
+ fake_clock_.AdvanceTimeMilliseconds(10);
+ }
+
+ event.Wait(/*ms=*/1000);
+ EXPECT_EQ(rtp_count, kExpected);
+
+ // Now stop send and yet feed more data.
+ egress_->StopSend();
+
+ // It should be safe to exit the test case while encoder_queue_ has
+ // outstanding data to process. We are making sure that this doesn't
+ // result in crahses or sanitizer errors due to remaining data.
+ for (size_t i = 0; i < kExpected * 2; i++) {
+ egress_->SendAudioData(GetAudioFrame(i));
+ fake_clock_.AdvanceTimeMilliseconds(10);
+ }
+}
+
+TEST_F(AudioEgressTest, ChangeEncoderFromPcmuToOpus) {
+ absl::optional<SdpAudioFormat> pcmu = egress_->GetEncoderFormat();
+ EXPECT_TRUE(pcmu);
+ EXPECT_EQ(pcmu->clockrate_hz, kPcmuFormat.clockrate_hz);
+ EXPECT_EQ(pcmu->num_channels, kPcmuFormat.num_channels);
+
+ constexpr int kOpusPayload = 120;
+ const SdpAudioFormat kOpusFormat = {"opus", 48000, 2};
+
+ egress_->SetEncoder(kOpusPayload, kOpusFormat,
+ encoder_factory_->MakeAudioEncoder(
+ kOpusPayload, kOpusFormat, absl::nullopt));
+
+ absl::optional<SdpAudioFormat> opus = egress_->GetEncoderFormat();
+ EXPECT_TRUE(opus);
+ EXPECT_EQ(opus->clockrate_hz, kOpusFormat.clockrate_hz);
+ EXPECT_EQ(opus->num_channels, kOpusFormat.num_channels);
+}
+
+TEST_F(AudioEgressTest, SendDTMF) {
+ constexpr int kExpected = 7;
+ constexpr int kPayloadType = 100;
+ constexpr int kDurationMs = 100;
+ constexpr int kSampleRate = 8000;
+ constexpr int kEvent = 3;
+
+ egress_->RegisterTelephoneEventType(kPayloadType, kSampleRate);
+ // 100 ms duration will produce total 7 DTMF
+ // 1 @ 20 ms, 2 @ 40 ms, 3 @ 60 ms, 4 @ 80 ms
+ // 5, 6, 7 @ 100 ms (last one sends 3 dtmf)
+ egress_->SendTelephoneEvent(kEvent, kDurationMs);
+
+ rtc::Event event;
+ int dtmf_count = 0;
+ auto is_dtmf = [&](RtpPacketReceived& rtp) {
+ return (rtp.PayloadType() == kPayloadType &&
+ rtp.SequenceNumber() == kSeqNum + dtmf_count &&
+ rtp.padding_size() == 0 && rtp.Marker() == (dtmf_count == 0) &&
+ rtp.Ssrc() == kRemoteSsrc);
+ };
+
+ // It's possible that we may have actual audio RTP packets along with
+ // DTMF packtets. We are only interested in the exact number of DTMF
+ // packets rtp stack is emitting.
+ auto rtp_sent = [&](const uint8_t* packet, size_t length, Unused) {
+ RtpPacketReceived rtp;
+ rtp.Parse(packet, length);
+ if (is_dtmf(rtp) && ++dtmf_count == kExpected) {
+ event.Set();
+ }
+ return true;
+ };
+
+ EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(rtp_sent));
+
+ // Two 10 ms audio frames will result in rtp packet with ptime 20.
+ for (size_t i = 0; i < kExpected * 2; i++) {
+ egress_->SendAudioData(GetAudioFrame(i));
+ fake_clock_.AdvanceTimeMilliseconds(10);
+ }
+
+ event.Wait(/*ms=*/1000);
+ EXPECT_EQ(dtmf_count, kExpected);
+}
+
+} // namespace
+} // namespace webrtc
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