Added RTCMediaStreamTrackStats.jitterBufferDelay for audio
Description of this stat can be found here:
https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-jitterbufferdelay
Bug: webrtc:8281
Change-Id: Ib2e8174f3449e68ad419ae2d58d5565fc9854023
Reviewed-on: https://webrtc-review.googlesource.com/3381
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20069}diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 5b92217..9dd60eb 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -522,6 +522,7 @@
NetEqDecodingTestFaxMode() : NetEqDecodingTest() {
config_.playout_mode = kPlayoutFax;
}
+ void TestJitterBufferDelay(bool apply_packet_loss);
};
TEST_F(NetEqDecodingTestFaxMode, TestFrameWaitingTimeStatistics) {
@@ -1684,4 +1685,64 @@
EXPECT_EQ(kNumConcealmentEvents, static_cast<int>(stats.concealment_events));
}
+// Test that the jitter buffer delay stat is computed correctly.
+void NetEqDecodingTestFaxMode::TestJitterBufferDelay(bool apply_packet_loss) {
+ const int kNumPackets = 10;
+ const int kDelayInNumPackets = 2;
+ const int kPacketLenMs = 10; // All packets are of 10 ms size.
+ const size_t kSamples = kPacketLenMs * 16;
+ const size_t kPayloadBytes = kSamples * 2;
+ RTPHeader rtp_info;
+ rtp_info.ssrc = 0x1234; // Just an arbitrary SSRC.
+ rtp_info.payloadType = 94; // PCM16b WB codec.
+ rtp_info.markerBit = 0;
+ const uint8_t payload[kPayloadBytes] = {0};
+ bool muted;
+ int packets_sent = 0;
+ int packets_received = 0;
+ int expected_delay = 0;
+ while (packets_received < kNumPackets) {
+ // Insert packet.
+ if (packets_sent < kNumPackets) {
+ rtp_info.sequenceNumber = packets_sent++;
+ rtp_info.timestamp = rtp_info.sequenceNumber * kSamples;
+ neteq_->InsertPacket(rtp_info, payload, 0);
+ }
+
+ // Get packet.
+ if (packets_sent > kDelayInNumPackets) {
+ neteq_->GetAudio(&out_frame_, &muted);
+ packets_received++;
+
+ // The delay reported by the jitter buffer never exceeds
+ // the number of samples previously fetched with GetAudio
+ // (hence the min()).
+ int packets_delay = std::min(packets_received, kDelayInNumPackets + 1);
+
+ // The increase of the expected delay is the product of
+ // the current delay of the jitter buffer in ms * the
+ // number of samples that are sent for play out.
+ int current_delay_ms = packets_delay * kPacketLenMs;
+ expected_delay += current_delay_ms * kSamples;
+ }
+ }
+
+ if (apply_packet_loss) {
+ // Extra call to GetAudio to cause concealment.
+ neteq_->GetAudio(&out_frame_, &muted);
+ }
+
+ // Check jitter buffer delay.
+ NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
+ EXPECT_EQ(expected_delay, static_cast<int>(stats.jitter_buffer_delay_ms));
+}
+
+TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithoutLoss) {
+ TestJitterBufferDelay(false);
+}
+
+TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithLoss) {
+ TestJitterBufferDelay(true);
+}
+
} // namespace webrtc