Adding upstream version 110.0.1.upstream/110.0.1upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 115 insertions, 0 deletions

diff --git a/third_party/libwebrtc/audio/test/audio_stats_test.cc b/third_party/libwebrtc/audio/test/audio_stats_test.cc
new file mode 100644
index 0000000000..febcb066fd
--- /dev/null
+++ b/third_party/libwebrtc/audio/test/audio_stats_test.cc
@@ -0,0 +1,115 @@
+/*
+ *  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 "audio/test/audio_end_to_end_test.h"
+#include "rtc_base/numerics/safe_compare.h"
+#include "system_wrappers/include/sleep.h"
+#include "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;
+
+  BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() const override {
+    BuiltInNetworkBehaviorConfig 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.payload_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_FALSE(send_stats.apm_statistics.delay_median_ms);
+    EXPECT_FALSE(send_stats.apm_statistics.delay_standard_deviation_ms);
+    EXPECT_FALSE(send_stats.apm_statistics.echo_return_loss);
+    EXPECT_FALSE(send_stats.apm_statistics.echo_return_loss_enhancement);
+    EXPECT_FALSE(send_stats.apm_statistics.residual_echo_likelihood);
+    EXPECT_FALSE(send_stats.apm_statistics.residual_echo_likelihood_recent_max);
+
+    AudioReceiveStreamInterface::Stats recv_stats =
+        receive_stream()->GetStats(/*get_and_clear_legacy_stats=*/true);
+    EXPECT_PRED2(IsNear, kBytesSent, recv_stats.payload_bytes_rcvd);
+    EXPECT_PRED2(IsNear, kPacketsSent, recv_stats.packets_rcvd);
+    EXPECT_EQ(0u, recv_stats.packets_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);
+  }
+};
+}  // namespace
+
+using AudioStatsTest = CallTest;
+
+TEST_F(AudioStatsTest, DISABLED_NoLoss) {
+  NoLossTest test;
+  RunBaseTest(&test);
+}
+
+}  // namespace test
+}  // namespace webrtc