Replace voe_conference_test.

webrtc/audio/test/audio_stats_test.cc

+/*
+ *  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 "webrtc/audio/test/audio_end_to_end_test.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+bool IsNear(int reference, int v) {
+  // Margin is 10%.
+  int error = reference / 10 + 1;

[kwiberg-webrtc, 2017/09/13 14:20:26]

const?

[the sun, 2017/09/13 14:53:33]

Done.

+  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;

[kwiberg-webrtc, 2017/09/13 14:20:26]

static constexpr? Or maybe just constexpr, if you don't want to deal with
defining them again outside the class...

[the sun, 2017/09/13 14:53:32]

constexpr on its own yields compiler errors:

../../webrtc/audio/test/audio_stats_test.cc:27:3: error: non-static data member
cannot be constexpr; did you intend to make it static?
  constexpr int kTestDurationMs = 8000;
  ^
  static

and static constexpr yields a link errors:

Undefined symbols for architecture x86_64:
  "webrtc::test::(anonymous namespace)::NoLossTest::kBytesSent", referenced
from:
      webrtc::test::(anonymous namespace)::NoLossTest::OnStreamsStopped() in
audio_stats_test.o

I guess I'd have to redeclare the static constexpr outside the class
declaration, but then simply using 'const' seemed less wordy...

[kwiberg-webrtc, 2017/09/14 01:26:40]

Yeah, that. You could use the enum hack (which got much more powerful in C++11)
or bite the bullet and make declarations outside the class body, but this is
just a test class, so we don't really care if we duplicate these constants in
each instance. Go with just "const".

(C++17 will finally solve this problem, by the way.)

+
+  NoLossTest() = default;
+
+  FakeNetworkPipe::Config GetNetworkPipeConfig() const override {
+    FakeNetworkPipe::Config 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);

[kwiberg-webrtc, 2017/09/13 14:20:26]

Should this comparison be case-insensitive? SDP codec names are case-insensitive
IIRC. Also, unlike all the other codecs, Opus's canonical name is lower-case...

[the sun, 2017/09/13 14:53:33]

Changing it to lower case, here and in call_test.cc

+    // 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_EQ(-1.0f, send_stats.aec_quality_min);
+    EXPECT_EQ(-1, send_stats.echo_delay_median_ms);
+    EXPECT_EQ(-1, send_stats.echo_delay_std_ms);
+    EXPECT_EQ(-100, send_stats.echo_return_loss);
+    EXPECT_EQ(-100, send_stats.echo_return_loss_enhancement);
+    EXPECT_EQ(0.0f, send_stats.residual_echo_likelihood);
+    EXPECT_EQ(0.0f, send_stats.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_EQ(send_stats.ext_seqnum,
+              static_cast<int32_t>(recv_stats.ext_seqnum));

[kwiberg-webrtc, 2017/09/13 14:20:26]

static_cast is dangerous. Can you convert both of them to a larger type instead,
using e.g. int64_t{x}? Otherwise, I suggest checked_cast, or SafeEq.

[the sun, 2017/09/13 14:53:33]

Done.

+  }
+};
+}  // namespace
+
+using AudioStatsTest = CallTest;
+
+TEST_F(AudioStatsTest, NoLoss) {
+  NoLossTest test;
+  RunBaseTest(&test);
+}
+
+}  // namespace test
+}  // namespace webrtc