| /* |
| * Copyright (c) 2017 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/test/audio_end_to_end_test.h" |
| #include "rtc_base/numerics/safe_compare.h" |
| #include "system_wrappers/include/sleep.h" |
| #include "test/gtest.h" |
| |
| namespace webrtc { |
| namespace test { |
| namespace { |
| |
| bool IsNear(int reference, int v) { |
| // Margin is 10%. |
| const int error = reference / 10 + 1; |
| return std::abs(reference - v) <= error; |
| } |
| |
| class NoLossTest : public AudioEndToEndTest { |
| public: |
| const int kTestDurationMs = 8000; |
| const int kBytesSent = 69351; |
| const int32_t kPacketsSent = 400; |
| const int64_t kRttMs = 100; |
| |
| NoLossTest() = default; |
| |
| DefaultNetworkSimulationConfig GetNetworkPipeConfig() const override { |
| DefaultNetworkSimulationConfig pipe_config; |
| pipe_config.queue_delay_ms = kRttMs / 2; |
| return pipe_config; |
| } |
| |
| void PerformTest() override { |
| SleepMs(kTestDurationMs); |
| send_audio_device()->StopRecording(); |
| AudioEndToEndTest::PerformTest(); |
| } |
| |
| void OnStreamsStopped() override { |
| AudioSendStream::Stats send_stats = send_stream()->GetStats(); |
| EXPECT_PRED2(IsNear, kBytesSent, send_stats.bytes_sent); |
| EXPECT_PRED2(IsNear, kPacketsSent, send_stats.packets_sent); |
| EXPECT_EQ(0, send_stats.packets_lost); |
| EXPECT_EQ(0.0f, send_stats.fraction_lost); |
| EXPECT_EQ("opus", send_stats.codec_name); |
| // send_stats.jitter_ms |
| EXPECT_PRED2(IsNear, kRttMs, send_stats.rtt_ms); |
| // Send level is 0 because it is cleared in TransmitMixer::StopSend(). |
| EXPECT_EQ(0, send_stats.audio_level); |
| // send_stats.total_input_energy |
| // send_stats.total_input_duration |
| EXPECT_FALSE(send_stats.apm_statistics.delay_median_ms); |
| EXPECT_FALSE(send_stats.apm_statistics.delay_standard_deviation_ms); |
| EXPECT_FALSE(send_stats.apm_statistics.echo_return_loss); |
| EXPECT_FALSE(send_stats.apm_statistics.echo_return_loss_enhancement); |
| EXPECT_FALSE(send_stats.apm_statistics.residual_echo_likelihood); |
| EXPECT_FALSE(send_stats.apm_statistics.residual_echo_likelihood_recent_max); |
| EXPECT_EQ(false, send_stats.typing_noise_detected); |
| |
| AudioReceiveStream::Stats recv_stats = receive_stream()->GetStats(); |
| EXPECT_PRED2(IsNear, kBytesSent, recv_stats.bytes_rcvd); |
| EXPECT_PRED2(IsNear, kPacketsSent, recv_stats.packets_rcvd); |
| EXPECT_EQ(0u, recv_stats.packets_lost); |
| EXPECT_EQ(0.0f, recv_stats.fraction_lost); |
| EXPECT_EQ("opus", send_stats.codec_name); |
| // recv_stats.jitter_ms |
| // recv_stats.jitter_buffer_ms |
| EXPECT_EQ(20u, recv_stats.jitter_buffer_preferred_ms); |
| // recv_stats.delay_estimate_ms |
| // Receive level is 0 because it is cleared in Channel::StopPlayout(). |
| EXPECT_EQ(0, recv_stats.audio_level); |
| // recv_stats.total_output_energy |
| // recv_stats.total_samples_received |
| // recv_stats.total_output_duration |
| // recv_stats.concealed_samples |
| // recv_stats.expand_rate |
| // recv_stats.speech_expand_rate |
| EXPECT_EQ(0.0, recv_stats.secondary_decoded_rate); |
| EXPECT_EQ(0.0, recv_stats.secondary_discarded_rate); |
| EXPECT_EQ(0.0, recv_stats.accelerate_rate); |
| EXPECT_EQ(0.0, recv_stats.preemptive_expand_rate); |
| EXPECT_EQ(0, recv_stats.decoding_calls_to_silence_generator); |
| // recv_stats.decoding_calls_to_neteq |
| // recv_stats.decoding_normal |
| // recv_stats.decoding_plc |
| EXPECT_EQ(0, recv_stats.decoding_cng); |
| // recv_stats.decoding_plc_cng |
| // recv_stats.decoding_muted_output |
| // Capture start time is -1 because we do not have an associated send stream |
| // on the receiver side. |
| EXPECT_EQ(-1, recv_stats.capture_start_ntp_time_ms); |
| |
| // Match these stats between caller and receiver. |
| EXPECT_EQ(send_stats.local_ssrc, recv_stats.remote_ssrc); |
| EXPECT_EQ(*send_stats.codec_payload_type, *recv_stats.codec_payload_type); |
| EXPECT_TRUE(rtc::SafeEq(send_stats.ext_seqnum, recv_stats.ext_seqnum)); |
| } |
| }; |
| } // namespace |
| |
| using AudioStatsTest = CallTest; |
| |
| TEST_F(AudioStatsTest, DISABLED_NoLoss) { |
| NoLossTest test; |
| RunBaseTest(&test); |
| } |
| |
| } // namespace test |
| } // namespace webrtc |