Implement AudioReceiveStream::GetStats().

webrtc/audio/audio_receive_stream.cc

 /*
  *  Copyright (c) 2015 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 "webrtc/audio/audio_receive_stream.h"
 
 #include <string>
 
+#include "webrtc/audio/conversion.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/voice_engine/include/voe_codec.h"
+#include "webrtc/voice_engine/include/voe_neteq_stats.h"
+#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
+#include "webrtc/voice_engine/include/voe_video_sync.h"
+#include "webrtc/voice_engine/include/voe_volume_control.h"
 
 namespace webrtc {
 std::string AudioReceiveStream::Config::Rtp::ToString() const {
   std::stringstream ss;
   ss << "{remote_ssrc: " << remote_ssrc;
   ss << ", extensions: [";
   for (size_t i = 0; i < extensions.size(); ++i) {
     ss << extensions[i].ToString();
-    if (i != extensions.size() - 1)
+    if (i != extensions.size() - 1) {
       ss << ", ";
+    }
   }
   ss << ']';
   ss << '}';
   return ss.str();
 }
 
 std::string AudioReceiveStream::Config::ToString() const {
   std::stringstream ss;
   ss << "{rtp: " << rtp.ToString();
   ss << ", voe_channel_id: " << voe_channel_id;
-  if (!sync_group.empty())
+  if (!sync_group.empty()) {
     ss << ", sync_group: " << sync_group;
+  }
   ss << '}';
   return ss.str();
 }
 
 namespace internal {
 AudioReceiveStream::AudioReceiveStream(
       RemoteBitrateEstimator* remote_bitrate_estimator,
-      const webrtc::AudioReceiveStream::Config& config)
+      const webrtc::AudioReceiveStream::Config& config,
+      VoiceEngine* voice_engine)
     : remote_bitrate_estimator_(remote_bitrate_estimator),
       config_(config),
+      voice_engine_(voice_engine),
+      voe_base_(voice_engine),
       rtp_header_parser_(RtpHeaderParser::Create()) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
   RTC_DCHECK(config.voe_channel_id != -1);
   RTC_DCHECK(remote_bitrate_estimator_ != nullptr);
+  RTC_DCHECK(voice_engine_ != nullptr);
   RTC_DCHECK(rtp_header_parser_ != nullptr);
   for (const auto& ext : config.rtp.extensions) {
     // One-byte-extension local identifiers are in the range 1-14 inclusive.
     RTC_DCHECK_GE(ext.id, 1);
     RTC_DCHECK_LE(ext.id, 14);
     if (ext.name == RtpExtension::kAudioLevel) {
       RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAudioLevel, ext.id));
     } else if (ext.name == RtpExtension::kAbsSendTime) {
       RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAbsoluteSendTime, ext.id));
     } else if (ext.name == RtpExtension::kTransportSequenceNumber) {
       RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionTransportSequenceNumber, ext.id));
     } else {
       RTC_NOTREACHED() << "Unsupported RTP extension.";
     }
   }
 }
 
 AudioReceiveStream::~AudioReceiveStream() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
 }
 
 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
-  return webrtc::AudioReceiveStream::Stats();
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  webrtc::AudioReceiveStream::Stats stats;
+  stats.remote_ssrc = config_.rtp.remote_ssrc;
+  ScopedVoEInterface<VoECodec> codec(voice_engine_);
+  ScopedVoEInterface<VoENetEqStats> neteq(voice_engine_);
+  ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine_);
+  ScopedVoEInterface<VoEVideoSync> sync(voice_engine_);
+  ScopedVoEInterface<VoEVolumeControl> volume(voice_engine_);
+  unsigned int ssrc = 0;
+  webrtc::CallStatistics cs = {0};
+  webrtc::CodecInst ci = {0};
+  // Only collect stats if we have seen some traffic with the SSRC.
+  if (rtp->GetRemoteSSRC(config_.voe_channel_id, ssrc) == -1 ||
+      rtp->GetRTCPStatistics(config_.voe_channel_id, cs) == -1 ||
+      codec->GetRecCodec(config_.voe_channel_id, ci) == -1) {
+    return stats;
+  }
+
+  stats.bytes_rcvd = cs.bytesReceived;
+  stats.packets_rcvd = cs.packetsReceived;
+  stats.packets_lost = cs.cumulativeLost;
+  stats.fraction_lost = static_cast<float>(cs.fractionLost) / (1 << 8);
+  if (ci.pltype != -1) {
+    stats.codec_name = ci.plname;
+  }
+
+  stats.ext_seqnum = cs.extendedMax;
+  if (ci.plfreq / 1000 > 0) {
+    stats.jitter_ms = cs.jitterSamples / (ci.plfreq / 1000);
+  }
+  {
+    int jitter_buffer_delay_ms = 0;
+    int playout_buffer_delay_ms = 0;
+    sync->GetDelayEstimate(config_.voe_channel_id, &jitter_buffer_delay_ms,
+                           &playout_buffer_delay_ms);
+    stats.delay_estimate_ms =
+        jitter_buffer_delay_ms + playout_buffer_delay_ms;
+  }
+  {
+    unsigned int level = 0;
+    if (volume->GetSpeechOutputLevelFullRange(config_.voe_channel_id, level)
+        != -1) {
+      stats.audio_level = static_cast<int32_t>(level);
+    }
+  }
+
+  webrtc::NetworkStatistics ns = {0};
+  if (neteq->GetNetworkStatistics(config_.voe_channel_id, ns) != -1) {
+    // Get jitter buffer and total delay (alg + jitter + playout) stats.
+    stats.jitter_buffer_ms = ns.currentBufferSize;
+    stats.jitter_buffer_preferred_ms = ns.preferredBufferSize;
+    stats.expand_rate = Q14ToFloat(ns.currentExpandRate);
+    stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate);
+    stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate);
+    stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate);
+    stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate);
+  }
+
+  webrtc::AudioDecodingCallStats ds;
+  if (neteq->GetDecodingCallStatistics(config_.voe_channel_id, &ds) != -1) {
+    stats.decoding_calls_to_silence_generator =
+        ds.calls_to_silence_generator;
+    stats.decoding_calls_to_neteq = ds.calls_to_neteq;
+    stats.decoding_normal = ds.decoded_normal;
+    stats.decoding_plc = ds.decoded_plc;
+    stats.decoding_cng = ds.decoded_cng;
+    stats.decoding_plc_cng = ds.decoded_plc_cng;
+  }
+
+  stats.capture_start_ntp_time_ms = cs.capture_start_ntp_time_ms_;
+
+  return stats;
 }
 
 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return config_;
 }
 
 void AudioReceiveStream::Start() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
 void AudioReceiveStream::Stop() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
 void AudioReceiveStream::SignalNetworkState(NetworkState state) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
 bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+  // TODO(solenberg): Tests call this function on a network thread, libjingle
+  // calls on the worker thread. We should move towards always using a network
+  // thread. Then this check can be enabled.
+  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
   return false;
 }
 
 bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
                                     size_t length,
                                     const PacketTime& packet_time) {
+  // TODO(solenberg): Tests call this function on a network thread, libjingle
+  // calls on the worker thread. We should move towards always using a network
+  // thread. Then this check can be enabled.
+  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
   RTPHeader header;
 
   if (!rtp_header_parser_->Parse(packet, length, &header)) {
     return false;
   }
 
   // Only forward if the parsed header has absolute sender time. RTP timestamps
   // may have different rates for audio and video and shouldn't be mixed.
   if (config_.combined_audio_video_bwe &&
       header.extension.hasAbsoluteSendTime) {
     int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
     if (packet_time.timestamp >= 0)
       arrival_time_ms = (packet_time.timestamp + 500) / 1000;
     size_t payload_size = length - header.headerLength;
     remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
                                               header, false);
   }
   return true;
 }
 }  // namespace internal
 }  // namespace webrtc