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46c4e60939
Introduce SimulatedNetworkReceiverInterface and switch DirectTransport on this interface. Also switch part of related users on DefaultNetworkSimulationConfig. This two changes united into single CL to prevent work duplication. Most changes were done because of stop including fake_network_pipe.h into direct_transport.h, so splitting this into 2 CLs will require first fix all imports of fake_network_pipe.h and then replace them on new API imports again. Bug: webrtc:9630 Change-Id: I87d4a6ff1bab72d04a9871a40441f4fbe028f4e6 Reviewed-on: https://webrtc-review.googlesource.com/94762 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24336}
117 lines
4.3 KiB
C++
117 lines
4.3 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "audio/test/audio_end_to_end_test.h"
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#include "rtc_base/numerics/safe_compare.h"
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#include "system_wrappers/include/sleep.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace test {
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namespace {
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bool IsNear(int reference, int v) {
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// Margin is 10%.
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const int error = reference / 10 + 1;
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return std::abs(reference - v) <= error;
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}
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class NoLossTest : public AudioEndToEndTest {
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public:
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const int kTestDurationMs = 8000;
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const int kBytesSent = 69351;
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const int32_t kPacketsSent = 400;
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const int64_t kRttMs = 100;
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NoLossTest() = default;
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DefaultNetworkSimulationConfig GetNetworkPipeConfig() const override {
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DefaultNetworkSimulationConfig pipe_config;
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pipe_config.queue_delay_ms = kRttMs / 2;
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return pipe_config;
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}
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void PerformTest() override {
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SleepMs(kTestDurationMs);
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send_audio_device()->StopRecording();
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AudioEndToEndTest::PerformTest();
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}
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void OnStreamsStopped() override {
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AudioSendStream::Stats send_stats = send_stream()->GetStats();
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EXPECT_PRED2(IsNear, kBytesSent, send_stats.bytes_sent);
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EXPECT_PRED2(IsNear, kPacketsSent, send_stats.packets_sent);
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EXPECT_EQ(0, send_stats.packets_lost);
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EXPECT_EQ(0.0f, send_stats.fraction_lost);
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EXPECT_EQ("opus", send_stats.codec_name);
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// send_stats.jitter_ms
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EXPECT_PRED2(IsNear, kRttMs, send_stats.rtt_ms);
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// Send level is 0 because it is cleared in TransmitMixer::StopSend().
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EXPECT_EQ(0, send_stats.audio_level);
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// send_stats.total_input_energy
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// send_stats.total_input_duration
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EXPECT_FALSE(send_stats.apm_statistics.delay_median_ms);
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EXPECT_FALSE(send_stats.apm_statistics.delay_standard_deviation_ms);
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EXPECT_FALSE(send_stats.apm_statistics.echo_return_loss);
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EXPECT_FALSE(send_stats.apm_statistics.echo_return_loss_enhancement);
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EXPECT_FALSE(send_stats.apm_statistics.residual_echo_likelihood);
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EXPECT_FALSE(send_stats.apm_statistics.residual_echo_likelihood_recent_max);
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EXPECT_EQ(false, send_stats.typing_noise_detected);
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AudioReceiveStream::Stats recv_stats = receive_stream()->GetStats();
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EXPECT_PRED2(IsNear, kBytesSent, recv_stats.bytes_rcvd);
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EXPECT_PRED2(IsNear, kPacketsSent, recv_stats.packets_rcvd);
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EXPECT_EQ(0u, recv_stats.packets_lost);
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EXPECT_EQ(0.0f, recv_stats.fraction_lost);
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EXPECT_EQ("opus", send_stats.codec_name);
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// recv_stats.jitter_ms
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// recv_stats.jitter_buffer_ms
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EXPECT_EQ(20u, recv_stats.jitter_buffer_preferred_ms);
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// recv_stats.delay_estimate_ms
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// Receive level is 0 because it is cleared in Channel::StopPlayout().
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EXPECT_EQ(0, recv_stats.audio_level);
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// recv_stats.total_output_energy
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// recv_stats.total_samples_received
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// recv_stats.total_output_duration
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// recv_stats.concealed_samples
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// recv_stats.expand_rate
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// recv_stats.speech_expand_rate
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EXPECT_EQ(0.0, recv_stats.secondary_decoded_rate);
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EXPECT_EQ(0.0, recv_stats.secondary_discarded_rate);
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EXPECT_EQ(0.0, recv_stats.accelerate_rate);
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EXPECT_EQ(0.0, recv_stats.preemptive_expand_rate);
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EXPECT_EQ(0, recv_stats.decoding_calls_to_silence_generator);
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// recv_stats.decoding_calls_to_neteq
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// recv_stats.decoding_normal
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// recv_stats.decoding_plc
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EXPECT_EQ(0, recv_stats.decoding_cng);
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// recv_stats.decoding_plc_cng
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// recv_stats.decoding_muted_output
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// Capture start time is -1 because we do not have an associated send stream
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// on the receiver side.
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EXPECT_EQ(-1, recv_stats.capture_start_ntp_time_ms);
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// Match these stats between caller and receiver.
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EXPECT_EQ(send_stats.local_ssrc, recv_stats.remote_ssrc);
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EXPECT_EQ(*send_stats.codec_payload_type, *recv_stats.codec_payload_type);
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EXPECT_TRUE(rtc::SafeEq(send_stats.ext_seqnum, recv_stats.ext_seqnum));
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}
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};
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} // namespace
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using AudioStatsTest = CallTest;
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TEST_F(AudioStatsTest, DISABLED_NoLoss) {
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NoLossTest test;
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RunBaseTest(&test);
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}
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} // namespace test
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} // namespace webrtc
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