NetEq: Create an audio interruption metric
This CL adds a new metric to NetEq, which logs whenever a loss
concealment event has lasted longer than 150 ms (an "interruption").
The number of such events, as well as the sum length of them, is kept
in a SampleCounter, which can be queried at any time.
Any initial PLC at the beginning of a call, before the first packet is
decoded, is ignored.
Unit tests and piping to neteq_rtpplay are included.
Bug: webrtc:10549
Change-Id: I8a224a34254c47c74317617f420f6de997232d88
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132796
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27781}
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 00b3174..025ed69 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -689,6 +689,65 @@
}
}
+// This test verifies that audio interruption is not logged for the initial
+// PLC period before the first packet is deocoded.
+// TODO(henrik.lundin) Maybe move this test to neteq_network_stats_unittest.cc.
+TEST_F(NetEqImplTest, NoAudioInterruptionLoggedBeforeFirstDecode) {
+ UseNoMocks();
+ CreateInstance();
+
+ const uint8_t kPayloadType = 17; // Just an arbitrary number.
+ const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
+ const int kSampleRateHz = 8000;
+ const size_t kPayloadLengthSamples =
+ static_cast<size_t>(10 * kSampleRateHz / 1000); // 10 ms.
+ const size_t kPayloadLengthBytes = kPayloadLengthSamples * 2;
+ uint8_t payload[kPayloadLengthBytes] = {0};
+ RTPHeader rtp_header;
+ rtp_header.payloadType = kPayloadType;
+ rtp_header.sequenceNumber = 0x1234;
+ rtp_header.timestamp = 0x12345678;
+ rtp_header.ssrc = 0x87654321;
+
+ // Register the payload type.
+ EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+ SdpAudioFormat("l16", 8000, 1)));
+
+ // Pull audio several times. No packets have been inserted yet.
+ const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+ AudioFrame output;
+ bool muted;
+ for (int i = 0; i < 100; ++i) {
+ EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+ ASSERT_LE(output.samples_per_channel_, kMaxOutputSize);
+ EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+ EXPECT_EQ(1u, output.num_channels_);
+ EXPECT_NE(AudioFrame::kNormalSpeech, output.speech_type_);
+ }
+
+ // Insert 10 packets.
+ for (size_t i = 0; i < 10; ++i) {
+ rtp_header.sequenceNumber++;
+ rtp_header.timestamp += kPayloadLengthSamples;
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(i + 1, packet_buffer_->NumPacketsInBuffer());
+ }
+
+ // Pull audio repeatedly and make sure we get normal output, that is not PLC.
+ for (size_t i = 0; i < 3; ++i) {
+ EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+ ASSERT_LE(output.samples_per_channel_, kMaxOutputSize);
+ EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+ EXPECT_EQ(1u, output.num_channels_);
+ EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_)
+ << "NetEq did not decode the packets as expected.";
+ }
+
+ auto lifetime_stats = neteq_->GetLifetimeStatistics();
+ EXPECT_EQ(0u, lifetime_stats.interruption_count);
+}
+
// This test verifies that NetEq can handle comfort noise and enters/quits codec
// internal CNG mode properly.
TEST_F(NetEqImplTest, CodecInternalCng) {