56d460902e
The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM. Bug: webrtc:8563, b/67926135 Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24 Reviewed-on: https://webrtc-review.googlesource.com/25621 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20877}
118 lines
4.3 KiB
C++
118 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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FakeNetworkPipe::Config GetNetworkPipeConfig() const override {
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FakeNetworkPipe::Config 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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