NetEq/ACM: Refactor how packet waiting times are calculated
With this change, the aggregates for packet waiting times are
calculated in NetEq's StatisticsCalculator insead of in
AcmReceiver. This simplifies things somewhat, and avoids having to
copy the raw data on polling.
R=ivoc@webrtc.org, minyue@webrtc.org
Review URL: https://codereview.webrtc.org/1296633002 .
Cr-Commit-Position: refs/heads/master@{#9778}
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index 03fde53..0c43024 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -507,46 +507,23 @@
ASSERT_EQ(kBlockSize16kHz, out_len);
}
- std::vector<int> waiting_times;
- neteq_->WaitingTimes(&waiting_times);
- EXPECT_EQ(num_frames, waiting_times.size());
+ NetEqNetworkStatistics stats;
+ EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
// Since all frames are dumped into NetEQ at once, but pulled out with 10 ms
// spacing (per definition), we expect the delay to increase with 10 ms for
- // each packet.
- for (size_t i = 0; i < waiting_times.size(); ++i) {
- EXPECT_EQ(static_cast<int>(i + 1) * 10, waiting_times[i]);
- }
+ // each packet. Thus, we are calculating the statistics for a series from 10
+ // to 300, in steps of 10 ms.
+ EXPECT_EQ(155, stats.mean_waiting_time_ms);
+ EXPECT_EQ(155, stats.median_waiting_time_ms);
+ EXPECT_EQ(10, stats.min_waiting_time_ms);
+ EXPECT_EQ(300, stats.max_waiting_time_ms);
// Check statistics again and make sure it's been reset.
- neteq_->WaitingTimes(&waiting_times);
- int len = waiting_times.size();
- EXPECT_EQ(0, len);
-
- // Process > 100 frames, and make sure that that we get statistics
- // only for 100 frames. Note the new SSRC, causing NetEQ to reset.
- num_frames = 110;
- for (size_t i = 0; i < num_frames; ++i) {
- uint16_t payload[kSamples] = {0};
- WebRtcRTPHeader rtp_info;
- rtp_info.header.sequenceNumber = i;
- rtp_info.header.timestamp = i * kSamples;
- rtp_info.header.ssrc = 0x1235; // Just an arbitrary SSRC.
- rtp_info.header.payloadType = 94; // PCM16b WB codec.
- rtp_info.header.markerBit = 0;
- ASSERT_EQ(0, neteq_->InsertPacket(
- rtp_info,
- reinterpret_cast<uint8_t*>(payload),
- kPayloadBytes, 0));
- size_t out_len;
- int num_channels;
- NetEqOutputType type;
- ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
- &num_channels, &type));
- ASSERT_EQ(kBlockSize16kHz, out_len);
- }
-
- neteq_->WaitingTimes(&waiting_times);
- EXPECT_EQ(100u, waiting_times.size());
+ EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
+ EXPECT_EQ(-1, stats.mean_waiting_time_ms);
+ EXPECT_EQ(-1, stats.median_waiting_time_ms);
+ EXPECT_EQ(-1, stats.min_waiting_time_ms);
+ EXPECT_EQ(-1, stats.max_waiting_time_ms);
}
TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimeNegative) {