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Unified Diff: webrtc/audio/test/audio_stats_test.cc

Issue 3008273002: Replace voe_conference_test. (Closed)
Patch Set: rebase Created 3 years, 3 months ago
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Index: webrtc/audio/test/audio_stats_test.cc
diff --git a/webrtc/audio/test/audio_stats_test.cc b/webrtc/audio/test/audio_stats_test.cc
new file mode 100644
index 0000000000000000000000000000000000000000..57dfbed758f9d73031288a9c830cae6163fcb130
--- /dev/null
+++ b/webrtc/audio/test/audio_stats_test.cc
@@ -0,0 +1,118 @@
+/*
+ * 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/rtc_base/safe_compare.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%.
+ 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;
+
+ 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);
+ // 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_TRUE(rtc::SafeEq(send_stats.ext_seqnum, recv_stats.ext_seqnum));
+ }
+};
+} // namespace
+
+using AudioStatsTest = CallTest;
+
+TEST_F(AudioStatsTest, NoLoss) {
+ NoLossTest test;
+ RunBaseTest(&test);
+}
+
+} // namespace test
+} // namespace webrtc
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