VoipStatistics subAPI and implementation.

- Adding an interface that fetches lifetime NetEq statistics struct.

Bug: webrtc:11989
Change-Id: I871455bccdd53a33dd260f744e03ec81d29fbfd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190200
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32516}

api/voip/BUILD.gn

@@ -16,10 +16,12 @@ rtc_source_set("voip_api") {
     "voip_dtmf.h",
     "voip_engine.h",
     "voip_network.h",
+    "voip_statistics.h",
   ]
   deps = [
     "..:array_view",
     "../audio_codecs:audio_codecs_api",
+    "../neteq:neteq_api",
   ]
   absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }

api/voip/voip_engine.h

@@ -17,6 +17,7 @@ class VoipBase;
 class VoipCodec;
 class VoipNetwork;
 class VoipDtmf;
+class VoipStatistics;
 
 // VoipEngine is the main interface serving as the entry point for all VoIP
 // APIs. A single instance of VoipEngine should suffice the most of the need for
@@ -84,6 +85,10 @@ class VoipEngine {
 
   // VoipDtmf provides DTMF event APIs to register and send DTMF events.
   virtual VoipDtmf& Dtmf() = 0;
+
+  // VoipStatistics provides performance metrics around audio decoding module
+  // and jitter buffer (NetEq).
+  virtual VoipStatistics& Statistics() = 0;
 };
 
 }  // namespace webrtc

api/voip/voip_statistics.h

@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2020 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.
+ */
+
+#ifndef API_VOIP_VOIP_STATISTICS_H_
+#define API_VOIP_VOIP_STATISTICS_H_
+
+#include "api/neteq/neteq.h"
+#include "api/voip/voip_base.h"
+
+namespace webrtc {
+
+// VoipStatistics interface provides the interfaces for querying metrics around
+// the jitter buffer (NetEq) performance.
+class VoipStatistics {
+ public:
+  // Gets the statistics from NetEq. Returns absl::nullopt when channel_id is
+  // invalid.
+  virtual absl::optional<NetEqLifetimeStatistics> GetNetEqStatistics(
+      ChannelId channel_id) = 0;
+
+ protected:
+  virtual ~VoipStatistics() = default;
+};
+
+}  // namespace webrtc
+
+#endif  // API_VOIP_VOIP_STATISTICS_H_

audio/voip/audio_channel.cc

@@ -129,4 +129,29 @@ void AudioChannel::StopPlay() {
   }
 }
 
+NetEqLifetimeStatistics AudioChannel::GetNetEqStatistics() {
+  NetEqLifetimeStatistics neteq_stats;
+  NetworkStatistics stats = ingress_->GetNetworkStatistics();
+  neteq_stats.total_samples_received = stats.totalSamplesReceived;
+  neteq_stats.concealed_samples = stats.concealedSamples;
+  neteq_stats.concealment_events = stats.concealmentEvents;
+  neteq_stats.jitter_buffer_delay_ms = stats.jitterBufferDelayMs;
+  neteq_stats.jitter_buffer_emitted_count = stats.jitterBufferEmittedCount;
+  neteq_stats.jitter_buffer_target_delay_ms = stats.jitterBufferTargetDelayMs;
+  neteq_stats.inserted_samples_for_deceleration =
+      stats.insertedSamplesForDeceleration;
+  neteq_stats.removed_samples_for_acceleration =
+      stats.removedSamplesForAcceleration;
+  neteq_stats.silent_concealed_samples = stats.silentConcealedSamples;
+  neteq_stats.fec_packets_received = stats.fecPacketsReceived;
+  neteq_stats.fec_packets_discarded = stats.fecPacketsDiscarded;
+  neteq_stats.delayed_packet_outage_samples = stats.delayedPacketOutageSamples;
+  neteq_stats.relative_packet_arrival_delay_ms =
+      stats.relativePacketArrivalDelayMs;
+  neteq_stats.interruption_count = stats.interruptionCount;
+  neteq_stats.total_interruption_duration_ms =
+      stats.totalInterruptionDurationMs;
+  return neteq_stats;
+}
+
 }  // namespace webrtc

audio/voip/audio_channel.h

@@ -18,6 +18,7 @@
 
 #include "api/task_queue/task_queue_factory.h"
 #include "api/voip/voip_base.h"
+#include "api/voip/voip_statistics.h"
 #include "audio/voip/audio_egress.h"
 #include "audio/voip/audio_ingress.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
@@ -81,6 +82,7 @@ class AudioChannel : public rtc::RefCountInterface {
   void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) {
     ingress_->SetReceiveCodecs(codecs);
   }
+  NetEqLifetimeStatistics GetNetEqStatistics();
 
  private:
   // ChannelId that this audio channel belongs for logging purpose.

audio/voip/audio_ingress.h

@@ -82,12 +82,8 @@ class AudioIngress : public AudioMixer::Source {
 
   NetworkStatistics GetNetworkStatistics() const {
     NetworkStatistics stats;
-    acm_receiver_.GetNetworkStatistics(&stats);
-    return stats;
-  }
-  AudioDecodingCallStats GetDecodingStatistics() const {
-    AudioDecodingCallStats stats;
-    acm_receiver_.GetDecodingCallStatistics(&stats);
+    acm_receiver_.GetNetworkStatistics(&stats,
+                                       /*get_and_clear_legacy_stats=*/false);
     return stats;
   }
 

audio/voip/test/audio_channel_unittest.cc

@@ -139,5 +139,36 @@ TEST_F(AudioChannelTest, VerifyLocalSsrcAsAssigned) {
   EXPECT_EQ(rtp.Ssrc(), kLocalSsrc);
 }
 
+// Check metrics after processing an RTP packet.
+TEST_F(AudioChannelTest, TestAudioStatistics) {
+  rtc::Event event;
+  auto loop_rtp = [&](const uint8_t* packet, size_t length, Unused) {
+    audio_channel_->ReceivedRTPPacket(
+        rtc::ArrayView<const uint8_t>(packet, length));
+    event.Set();
+    return true;
+  };
+  EXPECT_CALL(transport_, SendRtp).WillOnce(Invoke(loop_rtp));
+
+  auto audio_sender = audio_channel_->GetAudioSender();
+  audio_sender->SendAudioData(GetAudioFrame(0));
+  audio_sender->SendAudioData(GetAudioFrame(1));
+
+  event.Wait(/*give_up_after_ms=*/1000);
+
+  AudioFrame audio_frame;
+  audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
+
+  // Check a few fields as we wouldn't have enough samples verify most of them
+  // here.
+  absl::optional<NetEqLifetimeStatistics> neteq_stats =
+      audio_channel_->GetNetEqStatistics();
+  EXPECT_TRUE(neteq_stats);
+  EXPECT_EQ(neteq_stats->total_samples_received, 80ULL);
+  EXPECT_EQ(neteq_stats->concealed_samples, 0ULL);
+  EXPECT_EQ(neteq_stats->jitter_buffer_delay_ms, 1600ULL);
+  EXPECT_EQ(neteq_stats->interruption_count, 0);
+}
+
 }  // namespace
 }  // namespace webrtc

audio/voip/voip_core.cc

@@ -382,4 +382,12 @@ bool VoipCore::SendDtmfEvent(ChannelId channel,
   return false;
 }
 
+absl::optional<NetEqLifetimeStatistics> VoipCore::GetNetEqStatistics(
+    ChannelId channel) {
+  if (auto audio_channel = GetChannel(channel)) {
+    return audio_channel->GetNetEqStatistics();
+  }
+  return absl::nullopt;
+}
+
 }  // namespace webrtc

audio/voip/voip_core.h

@@ -26,6 +26,7 @@
 #include "api/voip/voip_dtmf.h"
 #include "api/voip/voip_engine.h"
 #include "api/voip/voip_network.h"
+#include "api/voip/voip_statistics.h"
 #include "audio/audio_transport_impl.h"
 #include "audio/voip/audio_channel.h"
 #include "modules/audio_device/include/audio_device.h"
@@ -47,7 +48,8 @@ class VoipCore : public VoipEngine,
                  public VoipBase,
                  public VoipNetwork,
                  public VoipCodec,
-                 public VoipDtmf {
+                 public VoipDtmf,
+                 public VoipStatistics {
  public:
   ~VoipCore() override = default;
 
@@ -70,6 +72,7 @@ class VoipCore : public VoipEngine,
   VoipNetwork& Network() override { return *this; }
   VoipCodec& Codec() override { return *this; }
   VoipDtmf& Dtmf() override { return *this; }
+  VoipStatistics& Statistics() override { return *this; }
 
   // Implements VoipBase interfaces.
   absl::optional<ChannelId> CreateChannel(
@@ -103,6 +106,10 @@ class VoipCore : public VoipEngine,
                      DtmfEvent dtmf_event,
                      int duration_ms) override;
 
+  // Implements VoipStatistics interfaces.
+  absl::optional<NetEqLifetimeStatistics> GetNetEqStatistics(
+      ChannelId channel) override;
+
  private:
   // Fetches the corresponding AudioChannel assigned with given |channel|.
   // Returns nullptr if not found.