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ac0a4cbbd8
This is a reland of fbde32e596
The chromium problem should be fixed with
https://chromium-review.googlesource.com/c/chromium/src/+/1862437
Original change's description:
> Fix GetStats bytesSent/Received, wireup headerBytesSent/Received
>
> Changes the standard GetStats, legacy GetStats unchanged.
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> Bug: webrtc:10525
> Change-Id: Ie10fe8079f1d8b4cc6bbe513f6a2fc91477b5441
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156084
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29462}
Tbr: kwiberg@webrtc.org
Bug: webrtc:10525
Change-Id: I3b61f9535aa3f1fca2ed84f068233803d4ec9fe2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157045
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29485}
115 lines
4.2 KiB
C++
115 lines
4.2 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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BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() const override {
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BuiltInNetworkBehaviorConfig 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.payload_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.payload_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("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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}
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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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