diff options
Diffstat (limited to 'third_party/libwebrtc/audio/voip/test')
6 files changed, 1271 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/voip/test/BUILD.gn b/third_party/libwebrtc/audio/voip/test/BUILD.gn new file mode 100644 index 0000000000..e89f2b001a --- /dev/null +++ b/third_party/libwebrtc/audio/voip/test/BUILD.gn @@ -0,0 +1,101 @@ +# 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. + +import("../../../webrtc.gni") + +if (rtc_include_tests) { + rtc_source_set("mock_task_queue") { + testonly = true + visibility = [ "*" ] + sources = [ "mock_task_queue.h" ] + deps = [ + "../../../api/task_queue:task_queue", + "../../../api/task_queue/test:mock_task_queue_base", + "../../../test:test_support", + ] + } + + if (!build_with_chromium) { + rtc_library("voip_core_unittests") { + testonly = true + sources = [ "voip_core_unittest.cc" ] + deps = [ + "..:voip_core", + "../../../api/audio_codecs:builtin_audio_decoder_factory", + "../../../api/audio_codecs:builtin_audio_encoder_factory", + "../../../api/task_queue:default_task_queue_factory", + "../../../modules/audio_device:mock_audio_device", + "../../../modules/audio_processing:mocks", + "../../../test:audio_codec_mocks", + "../../../test:mock_transport", + "../../../test:run_loop", + "../../../test:test_support", + ] + } + } + + rtc_library("audio_channel_unittests") { + testonly = true + sources = [ "audio_channel_unittest.cc" ] + deps = [ + ":mock_task_queue", + "..:audio_channel", + "../../../api:transport_api", + "../../../api/audio_codecs:builtin_audio_decoder_factory", + "../../../api/audio_codecs:builtin_audio_encoder_factory", + "../../../api/task_queue:task_queue", + "../../../modules/audio_mixer:audio_mixer_impl", + "../../../modules/audio_mixer:audio_mixer_test_utils", + "../../../modules/rtp_rtcp:rtp_rtcp", + "../../../modules/rtp_rtcp:rtp_rtcp_format", + "../../../rtc_base:logging", + "../../../test:mock_transport", + "../../../test:test_support", + ] + absl_deps = [ "//third_party/abseil-cpp/absl/functional:any_invocable" ] + } + + rtc_library("audio_ingress_unittests") { + testonly = true + sources = [ "audio_ingress_unittest.cc" ] + deps = [ + "..:audio_egress", + "..:audio_ingress", + "../../../api:transport_api", + "../../../api/audio_codecs:builtin_audio_decoder_factory", + "../../../api/audio_codecs:builtin_audio_encoder_factory", + "../../../api/task_queue:default_task_queue_factory", + "../../../modules/audio_mixer:audio_mixer_test_utils", + "../../../modules/rtp_rtcp:rtp_rtcp", + "../../../rtc_base:logging", + "../../../rtc_base:rtc_event", + "../../../test:mock_transport", + "../../../test:run_loop", + "../../../test:test_support", + ] + } + + rtc_library("audio_egress_unittests") { + testonly = true + sources = [ "audio_egress_unittest.cc" ] + deps = [ + "..:audio_egress", + "../../../api:transport_api", + "../../../api/audio_codecs:builtin_audio_encoder_factory", + "../../../api/task_queue:default_task_queue_factory", + "../../../modules/audio_mixer:audio_mixer_test_utils", + "../../../modules/rtp_rtcp:rtp_rtcp", + "../../../modules/rtp_rtcp:rtp_rtcp_format", + "../../../rtc_base:logging", + "../../../rtc_base:rtc_event", + "../../../test:mock_transport", + "../../../test:run_loop", + "../../../test:test_support", + ] + } +} diff --git a/third_party/libwebrtc/audio/voip/test/audio_channel_unittest.cc b/third_party/libwebrtc/audio/voip/test/audio_channel_unittest.cc new file mode 100644 index 0000000000..8955810429 --- /dev/null +++ b/third_party/libwebrtc/audio/voip/test/audio_channel_unittest.cc @@ -0,0 +1,357 @@ +/* + * 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 diff --git a/third_party/libwebrtc/audio/voip/test/audio_egress_unittest.cc b/third_party/libwebrtc/audio/voip/test/audio_egress_unittest.cc new file mode 100644 index 0000000000..34c5585347 --- /dev/null +++ b/third_party/libwebrtc/audio/voip/test/audio_egress_unittest.cc @@ -0,0 +1,327 @@ +/* + * 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 "modules/rtp_rtcp/source/rtp_rtcp_impl2.h" +#include "rtc_base/event.h" +#include "rtc_base/logging.h" +#include "test/gmock.h" +#include "test/gtest.h" +#include "test/mock_transport.h" +#include "test/run_loop.h" + +namespace webrtc { +namespace { + +using ::testing::Invoke; +using ::testing::NiceMock; +using ::testing::Unused; + +std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpStack(Clock* clock, + Transport* transport, + uint32_t remote_ssrc) { + RtpRtcpInterface::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 = ModuleRtpRtcpImpl2::Create(rtp_config); + rtp_rtcp->SetSendingMediaStatus(false); + rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound); + return rtp_rtcp; +} + +constexpr int16_t kAudioLevel = 3004; // Used for sine wave level. + +// 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 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) { + 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 { + rtp_rtcp_ = CreateRtpStack(&fake_clock_, &transport_, kRemoteSsrc); + 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 { + egress_->StopSend(); + rtp_rtcp_->SetSendingStatus(false); + egress_.reset(); + rtp_rtcp_.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; + } + + test::RunLoop run_loop_; + // 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<ModuleRtpRtcpImpl2> rtp_rtcp_; + std::unique_ptr<TaskQueueFactory> task_queue_factory_; + rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_; + std::unique_ptr<AudioEgress> egress_; +}; + +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(TimeDelta::Seconds(1)); + 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(TimeDelta::Seconds(1)); + 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(TimeDelta::Seconds(1)); + 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(TimeDelta::Seconds(1)); + EXPECT_EQ(dtmf_count, kExpected); +} + +TEST_F(AudioEgressTest, TestAudioInputLevelAndEnergyDuration) { + // Per audio_level's kUpdateFrequency, we need more than 10 audio samples to + // get audio level from input source. + constexpr int kExpected = 6; + 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(/*give_up_after=*/TimeDelta::Seconds(1)); + EXPECT_EQ(rtp_count, kExpected); + + constexpr double kExpectedEnergy = 0.00016809565587789564; + constexpr double kExpectedDuration = 0.11999999999999998; + + EXPECT_EQ(egress_->GetInputAudioLevel(), kAudioLevel); + EXPECT_DOUBLE_EQ(egress_->GetInputTotalEnergy(), kExpectedEnergy); + EXPECT_DOUBLE_EQ(egress_->GetInputTotalDuration(), kExpectedDuration); +} + +} // namespace +} // namespace webrtc diff --git a/third_party/libwebrtc/audio/voip/test/audio_ingress_unittest.cc b/third_party/libwebrtc/audio/voip/test/audio_ingress_unittest.cc new file mode 100644 index 0000000000..3c309dbf82 --- /dev/null +++ b/third_party/libwebrtc/audio/voip/test/audio_ingress_unittest.cc @@ -0,0 +1,238 @@ +/* + * 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_ingress.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/default_task_queue_factory.h" +#include "audio/voip/audio_egress.h" +#include "modules/audio_mixer/sine_wave_generator.h" +#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h" +#include "rtc_base/event.h" +#include "rtc_base/logging.h" +#include "test/gmock.h" +#include "test/gtest.h" +#include "test/mock_transport.h" +#include "test/run_loop.h" + +namespace webrtc { +namespace { + +using ::testing::Invoke; +using ::testing::NiceMock; +using ::testing::Unused; + +constexpr int16_t kAudioLevel = 3004; // Used for sine wave level. + +class AudioIngressTest : public ::testing::Test { + public: + const SdpAudioFormat kPcmuFormat = {"pcmu", 8000, 1}; + + AudioIngressTest() + : fake_clock_(123456789), wave_generator_(1000.0, kAudioLevel) { + receive_statistics_ = ReceiveStatistics::Create(&fake_clock_); + + RtpRtcpInterface::Configuration rtp_config; + rtp_config.clock = &fake_clock_; + rtp_config.audio = true; + rtp_config.receive_statistics = receive_statistics_.get(); + rtp_config.rtcp_report_interval_ms = 5000; + rtp_config.outgoing_transport = &transport_; + rtp_config.local_media_ssrc = 0xdeadc0de; + rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(rtp_config); + + rtp_rtcp_->SetSendingMediaStatus(false); + rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound); + + task_queue_factory_ = CreateDefaultTaskQueueFactory(); + encoder_factory_ = CreateBuiltinAudioEncoderFactory(); + decoder_factory_ = CreateBuiltinAudioDecoderFactory(); + } + + void SetUp() override { + constexpr int kPcmuPayload = 0; + ingress_ = std::make_unique<AudioIngress>(rtp_rtcp_.get(), &fake_clock_, + receive_statistics_.get(), + decoder_factory_); + ingress_->SetReceiveCodecs({{kPcmuPayload, kPcmuFormat}}); + + egress_ = std::make_unique<AudioEgress>(rtp_rtcp_.get(), &fake_clock_, + task_queue_factory_.get()); + egress_->SetEncoder(kPcmuPayload, kPcmuFormat, + encoder_factory_->MakeAudioEncoder( + kPcmuPayload, kPcmuFormat, absl::nullopt)); + egress_->StartSend(); + ingress_->StartPlay(); + rtp_rtcp_->SetSendingStatus(true); + } + + void TearDown() override { + rtp_rtcp_->SetSendingStatus(false); + ingress_->StopPlay(); + egress_->StopSend(); + egress_.reset(); + ingress_.reset(); + } + + 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; + } + + test::RunLoop run_loop_; + SimulatedClock fake_clock_; + SineWaveGenerator wave_generator_; + NiceMock<MockTransport> transport_; + std::unique_ptr<ReceiveStatistics> receive_statistics_; + std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_; + rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_; + rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; + std::unique_ptr<TaskQueueFactory> task_queue_factory_; + std::unique_ptr<AudioIngress> ingress_; + std::unique_ptr<AudioEgress> egress_; +}; + +TEST_F(AudioIngressTest, PlayingAfterStartAndStop) { + EXPECT_EQ(ingress_->IsPlaying(), true); + ingress_->StopPlay(); + EXPECT_EQ(ingress_->IsPlaying(), false); +} + +TEST_F(AudioIngressTest, GetAudioFrameAfterRtpReceived) { + rtc::Event event; + auto handle_rtp = [&](const uint8_t* packet, size_t length, Unused) { + ingress_->ReceivedRTPPacket(rtc::ArrayView<const uint8_t>(packet, length)); + event.Set(); + return true; + }; + EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(handle_rtp)); + egress_->SendAudioData(GetAudioFrame(0)); + egress_->SendAudioData(GetAudioFrame(1)); + event.Wait(TimeDelta::Seconds(1)); + + AudioFrame audio_frame; + EXPECT_EQ( + ingress_->GetAudioFrameWithInfo(kPcmuFormat.clockrate_hz, &audio_frame), + AudioMixer::Source::AudioFrameInfo::kNormal); + EXPECT_FALSE(audio_frame.muted()); + EXPECT_EQ(audio_frame.num_channels_, 1u); + EXPECT_EQ(audio_frame.samples_per_channel_, + static_cast<size_t>(kPcmuFormat.clockrate_hz / 100)); + EXPECT_EQ(audio_frame.sample_rate_hz_, kPcmuFormat.clockrate_hz); + EXPECT_NE(audio_frame.timestamp_, 0u); + EXPECT_EQ(audio_frame.elapsed_time_ms_, 0); +} + +TEST_F(AudioIngressTest, TestSpeechOutputLevelAndEnergyDuration) { + // Per audio_level's kUpdateFrequency, we need more than 10 audio samples to + // get audio level from output source. + constexpr int kNumRtp = 6; + int rtp_count = 0; + rtc::Event event; + auto handle_rtp = [&](const uint8_t* packet, size_t length, Unused) { + ingress_->ReceivedRTPPacket(rtc::ArrayView<const uint8_t>(packet, length)); + if (++rtp_count == kNumRtp) { + event.Set(); + } + return true; + }; + EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(handle_rtp)); + for (int i = 0; i < kNumRtp * 2; i++) { + egress_->SendAudioData(GetAudioFrame(i)); + fake_clock_.AdvanceTimeMilliseconds(10); + } + event.Wait(/*give_up_after=*/TimeDelta::Seconds(1)); + + for (int i = 0; i < kNumRtp * 2; ++i) { + AudioFrame audio_frame; + EXPECT_EQ( + ingress_->GetAudioFrameWithInfo(kPcmuFormat.clockrate_hz, &audio_frame), + AudioMixer::Source::AudioFrameInfo::kNormal); + } + EXPECT_EQ(ingress_->GetOutputAudioLevel(), kAudioLevel); + + constexpr double kExpectedEnergy = 0.00016809565587789564; + constexpr double kExpectedDuration = 0.11999999999999998; + + EXPECT_DOUBLE_EQ(ingress_->GetOutputTotalEnergy(), kExpectedEnergy); + EXPECT_DOUBLE_EQ(ingress_->GetOutputTotalDuration(), kExpectedDuration); +} + +TEST_F(AudioIngressTest, PreferredSampleRate) { + rtc::Event event; + auto handle_rtp = [&](const uint8_t* packet, size_t length, Unused) { + ingress_->ReceivedRTPPacket(rtc::ArrayView<const uint8_t>(packet, length)); + event.Set(); + return true; + }; + EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(handle_rtp)); + egress_->SendAudioData(GetAudioFrame(0)); + egress_->SendAudioData(GetAudioFrame(1)); + event.Wait(TimeDelta::Seconds(1)); + + AudioFrame audio_frame; + EXPECT_EQ( + ingress_->GetAudioFrameWithInfo(kPcmuFormat.clockrate_hz, &audio_frame), + AudioMixer::Source::AudioFrameInfo::kNormal); + EXPECT_EQ(ingress_->PreferredSampleRate(), kPcmuFormat.clockrate_hz); +} + +// This test highlights the case where caller invokes StopPlay() which then +// AudioIngress should play silence frame afterwards. +TEST_F(AudioIngressTest, GetMutedAudioFrameAfterRtpReceivedAndStopPlay) { + // StopPlay before we start sending RTP packet with sine wave. + ingress_->StopPlay(); + + // Send 6 RTP packets to generate more than 100 ms audio sample to get + // valid speech level. + constexpr int kNumRtp = 6; + int rtp_count = 0; + rtc::Event event; + auto handle_rtp = [&](const uint8_t* packet, size_t length, Unused) { + ingress_->ReceivedRTPPacket(rtc::ArrayView<const uint8_t>(packet, length)); + if (++rtp_count == kNumRtp) { + event.Set(); + } + return true; + }; + EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(handle_rtp)); + for (int i = 0; i < kNumRtp * 2; i++) { + egress_->SendAudioData(GetAudioFrame(i)); + fake_clock_.AdvanceTimeMilliseconds(10); + } + event.Wait(/*give_up_after=*/TimeDelta::Seconds(1)); + + for (int i = 0; i < kNumRtp * 2; ++i) { + AudioFrame audio_frame; + EXPECT_EQ( + ingress_->GetAudioFrameWithInfo(kPcmuFormat.clockrate_hz, &audio_frame), + AudioMixer::Source::AudioFrameInfo::kMuted); + const int16_t* audio_data = audio_frame.data(); + size_t length = + audio_frame.samples_per_channel_ * audio_frame.num_channels_; + for (size_t j = 0; j < length; ++j) { + EXPECT_EQ(audio_data[j], 0); + } + } + + // Now we should still see valid speech output level as StopPlay won't affect + // the measurement. + EXPECT_EQ(ingress_->GetOutputAudioLevel(), kAudioLevel); +} + +} // namespace +} // namespace webrtc diff --git a/third_party/libwebrtc/audio/voip/test/mock_task_queue.h b/third_party/libwebrtc/audio/voip/test/mock_task_queue.h new file mode 100644 index 0000000000..547b0d3f75 --- /dev/null +++ b/third_party/libwebrtc/audio/voip/test/mock_task_queue.h @@ -0,0 +1,55 @@ +/* + * 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. + */ + +#ifndef AUDIO_VOIP_TEST_MOCK_TASK_QUEUE_H_ +#define AUDIO_VOIP_TEST_MOCK_TASK_QUEUE_H_ + +#include <memory> + +#include "api/task_queue/task_queue_factory.h" +#include "api/task_queue/test/mock_task_queue_base.h" +#include "test/gmock.h" + +namespace webrtc { + +// MockTaskQueue enables immediate task run from global TaskQueueBase. +// It's necessary for some tests depending on TaskQueueBase internally. +class MockTaskQueue : public MockTaskQueueBase { + public: + MockTaskQueue() : current_(this) {} + + // Delete is deliberately defined as no-op as MockTaskQueue is expected to + // hold onto current global TaskQueueBase throughout the testing. + void Delete() override {} + + private: + CurrentTaskQueueSetter current_; +}; + +class MockTaskQueueFactory : public TaskQueueFactory { + public: + explicit MockTaskQueueFactory(MockTaskQueue* task_queue) + : task_queue_(task_queue) {} + + std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateTaskQueue( + absl::string_view name, + Priority priority) const override { + // Default MockTaskQueue::Delete is no-op, therefore it's safe to pass the + // raw pointer. + return std::unique_ptr<TaskQueueBase, TaskQueueDeleter>(task_queue_); + } + + private: + MockTaskQueue* task_queue_; +}; + +} // namespace webrtc + +#endif // AUDIO_VOIP_TEST_MOCK_TASK_QUEUE_H_ diff --git a/third_party/libwebrtc/audio/voip/test/voip_core_unittest.cc b/third_party/libwebrtc/audio/voip/test/voip_core_unittest.cc new file mode 100644 index 0000000000..b432506b12 --- /dev/null +++ b/third_party/libwebrtc/audio/voip/test/voip_core_unittest.cc @@ -0,0 +1,193 @@ +/* + * 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/voip_core.h" + +#include "api/audio_codecs/builtin_audio_decoder_factory.h" +#include "api/audio_codecs/builtin_audio_encoder_factory.h" +#include "api/task_queue/default_task_queue_factory.h" +#include "modules/audio_device/include/mock_audio_device.h" +#include "modules/audio_processing/include/mock_audio_processing.h" +#include "test/gtest.h" +#include "test/mock_transport.h" +#include "test/run_loop.h" + +namespace webrtc { +namespace { + +using ::testing::NiceMock; +using ::testing::Return; + +constexpr int kPcmuPayload = 0; +constexpr int kPcmuSampleRateHz = 8000; +constexpr int kDtmfEventDurationMs = 1000; +constexpr DtmfEvent kDtmfEventCode = DtmfEvent::kDigitZero; + +class VoipCoreTest : public ::testing::Test { + public: + const SdpAudioFormat kPcmuFormat = {"pcmu", 8000, 1}; + + VoipCoreTest() { audio_device_ = test::MockAudioDeviceModule::CreateNice(); } + + void SetUp() override { + auto encoder_factory = CreateBuiltinAudioEncoderFactory(); + auto decoder_factory = CreateBuiltinAudioDecoderFactory(); + rtc::scoped_refptr<AudioProcessing> audio_processing = + rtc::make_ref_counted<NiceMock<test::MockAudioProcessing>>(); + + voip_core_ = std::make_unique<VoipCore>( + std::move(encoder_factory), std::move(decoder_factory), + CreateDefaultTaskQueueFactory(), audio_device_, + std::move(audio_processing)); + } + + test::RunLoop run_loop_; + std::unique_ptr<VoipCore> voip_core_; + NiceMock<MockTransport> transport_; + rtc::scoped_refptr<test::MockAudioDeviceModule> audio_device_; +}; + +// Validate expected API calls that involves with VoipCore. Some verification is +// involved with checking mock audio device. +TEST_F(VoipCoreTest, BasicVoipCoreOperation) { + // Program mock as non-operational and ready to start. + EXPECT_CALL(*audio_device_, Recording()).WillOnce(Return(false)); + EXPECT_CALL(*audio_device_, Playing()).WillOnce(Return(false)); + EXPECT_CALL(*audio_device_, InitRecording()).WillOnce(Return(0)); + EXPECT_CALL(*audio_device_, InitPlayout()).WillOnce(Return(0)); + EXPECT_CALL(*audio_device_, StartRecording()).WillOnce(Return(0)); + EXPECT_CALL(*audio_device_, StartPlayout()).WillOnce(Return(0)); + + auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de); + + EXPECT_EQ(voip_core_->SetSendCodec(channel, kPcmuPayload, kPcmuFormat), + VoipResult::kOk); + EXPECT_EQ( + voip_core_->SetReceiveCodecs(channel, {{kPcmuPayload, kPcmuFormat}}), + VoipResult::kOk); + + EXPECT_EQ(voip_core_->StartSend(channel), VoipResult::kOk); + EXPECT_EQ(voip_core_->StartPlayout(channel), VoipResult::kOk); + + EXPECT_EQ(voip_core_->RegisterTelephoneEventType(channel, kPcmuPayload, + kPcmuSampleRateHz), + VoipResult::kOk); + + EXPECT_EQ( + voip_core_->SendDtmfEvent(channel, kDtmfEventCode, kDtmfEventDurationMs), + VoipResult::kOk); + + // Program mock as operational that is ready to be stopped. + EXPECT_CALL(*audio_device_, Recording()).WillOnce(Return(true)); + EXPECT_CALL(*audio_device_, Playing()).WillOnce(Return(true)); + EXPECT_CALL(*audio_device_, StopRecording()).WillOnce(Return(0)); + EXPECT_CALL(*audio_device_, StopPlayout()).WillOnce(Return(0)); + + EXPECT_EQ(voip_core_->StopSend(channel), VoipResult::kOk); + EXPECT_EQ(voip_core_->StopPlayout(channel), VoipResult::kOk); + EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk); +} + +TEST_F(VoipCoreTest, ExpectFailToUseReleasedChannelId) { + auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de); + + // Release right after creation. + EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk); + + // Now use released channel. + + EXPECT_EQ(voip_core_->SetSendCodec(channel, kPcmuPayload, kPcmuFormat), + VoipResult::kInvalidArgument); + EXPECT_EQ( + voip_core_->SetReceiveCodecs(channel, {{kPcmuPayload, kPcmuFormat}}), + VoipResult::kInvalidArgument); + EXPECT_EQ(voip_core_->RegisterTelephoneEventType(channel, kPcmuPayload, + kPcmuSampleRateHz), + VoipResult::kInvalidArgument); + EXPECT_EQ(voip_core_->StartSend(channel), VoipResult::kInvalidArgument); + EXPECT_EQ(voip_core_->StartPlayout(channel), VoipResult::kInvalidArgument); + EXPECT_EQ( + voip_core_->SendDtmfEvent(channel, kDtmfEventCode, kDtmfEventDurationMs), + VoipResult::kInvalidArgument); +} + +TEST_F(VoipCoreTest, SendDtmfEventWithoutRegistering) { + // Program mock as non-operational and ready to start send. + EXPECT_CALL(*audio_device_, Recording()).WillOnce(Return(false)); + EXPECT_CALL(*audio_device_, InitRecording()).WillOnce(Return(0)); + EXPECT_CALL(*audio_device_, StartRecording()).WillOnce(Return(0)); + + auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de); + + EXPECT_EQ(voip_core_->SetSendCodec(channel, kPcmuPayload, kPcmuFormat), + VoipResult::kOk); + + EXPECT_EQ(voip_core_->StartSend(channel), VoipResult::kOk); + // Send Dtmf event without registering beforehand, thus payload + // type is not set and kFailedPrecondition is expected. + EXPECT_EQ( + voip_core_->SendDtmfEvent(channel, kDtmfEventCode, kDtmfEventDurationMs), + VoipResult::kFailedPrecondition); + + // Program mock as sending and is ready to be stopped. + EXPECT_CALL(*audio_device_, Recording()).WillOnce(Return(true)); + EXPECT_CALL(*audio_device_, StopRecording()).WillOnce(Return(0)); + + EXPECT_EQ(voip_core_->StopSend(channel), VoipResult::kOk); + EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk); +} + +TEST_F(VoipCoreTest, SendDtmfEventWithoutStartSend) { + auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de); + + EXPECT_EQ(voip_core_->RegisterTelephoneEventType(channel, kPcmuPayload, + kPcmuSampleRateHz), + VoipResult::kOk); + + // Send Dtmf event without calling StartSend beforehand, thus + // Dtmf events cannot be sent and kFailedPrecondition is expected. + EXPECT_EQ( + voip_core_->SendDtmfEvent(channel, kDtmfEventCode, kDtmfEventDurationMs), + VoipResult::kFailedPrecondition); + + EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk); +} + +TEST_F(VoipCoreTest, StartSendAndPlayoutWithoutSettingCodec) { + auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de); + + // Call StartSend and StartPlayout without setting send/receive + // codec. Code should see that codecs aren't set and return false. + EXPECT_EQ(voip_core_->StartSend(channel), VoipResult::kFailedPrecondition); + EXPECT_EQ(voip_core_->StartPlayout(channel), VoipResult::kFailedPrecondition); + + EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk); +} + +TEST_F(VoipCoreTest, StopSendAndPlayoutWithoutStarting) { + auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de); + + EXPECT_EQ(voip_core_->SetSendCodec(channel, kPcmuPayload, kPcmuFormat), + VoipResult::kOk); + EXPECT_EQ( + voip_core_->SetReceiveCodecs(channel, {{kPcmuPayload, kPcmuFormat}}), + VoipResult::kOk); + + // Call StopSend and StopPlayout without starting them in + // the first place. Should see that it is already in the + // stopped state and return true. + EXPECT_EQ(voip_core_->StopSend(channel), VoipResult::kOk); + EXPECT_EQ(voip_core_->StopPlayout(channel), VoipResult::kOk); + + EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk); +} + +} // namespace +} // namespace webrtc |