Implement AudioSendStream::GetStats().

webrtc/audio/audio_send_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_send_stream.h"
 
 #include <string>
 
+#include "webrtc/audio/conversion.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/voice_engine/include/voe_audio_processing.h"
+#include "webrtc/voice_engine/include/voe_codec.h"
+#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
+#include "webrtc/voice_engine/include/voe_volume_control.h"
 
 namespace webrtc {
 std::string AudioSendStream::Config::Rtp::ToString() const {
   std::stringstream ss;
   ss << "{ssrc: " << 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 AudioSendStream::Config::ToString() const {
   std::stringstream ss;
   ss << "{rtp: " << rtp.ToString();
   ss << ", voe_channel_id: " << voe_channel_id;
   // TODO(solenberg): Encoder config.
   ss << ", cng_payload_type: " << cng_payload_type;
   ss << ", red_payload_type: " << red_payload_type;
   ss << '}';
   return ss.str();
 }
 
 namespace internal {
-AudioSendStream::AudioSendStream(const webrtc::AudioSendStream::Config& config)
-    : config_(config) {
+AudioSendStream::AudioSendStream(const webrtc::AudioSendStream::Config& config,
+                                 VoiceEngine* voice_engine)
+    : config_(config),
+      voice_engine_(voice_engine),
+      voe_base_(voice_engine) {
   LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
   RTC_DCHECK(config.voe_channel_id != -1);
+  RTC_DCHECK(voice_engine_ != nullptr);

[kwiberg-webrtc, 2015/10/26 13:19:53]

Just RTC_DCHECK(voice_engine_);? Otherwise, RTC_DCHECK_NE.

[the sun, 2015/10/26 14:44:02]

Done.

 }
 
 AudioSendStream::~AudioSendStream() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
 }
 
 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
-  return webrtc::AudioSendStream::Stats();
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  webrtc::AudioSendStream::Stats stats;
+  stats.local_ssrc = config_.rtp.ssrc;
+  ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine_);
+  ScopedVoEInterface<VoECodec> codec(voice_engine_);
+  ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine_);
+  ScopedVoEInterface<VoEVolumeControl> volume(voice_engine_);
+  unsigned int ssrc = 0;
+  webrtc::CallStatistics call_stats = {0};
+  if (rtp->GetLocalSSRC(config_.voe_channel_id, ssrc) == -1 ||
+      rtp->GetRTCPStatistics(config_.voe_channel_id, call_stats) == -1) {
+    return stats;
+  }
+
+  stats.bytes_sent = call_stats.bytesSent;
+  stats.packets_sent = call_stats.packetsSent;
+
+  webrtc::CodecInst codec_inst = {0};
+  if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) {
+    RTC_DCHECK(codec_inst.pltype != -1);

[kwiberg-webrtc, 2015/10/26 13:19:53]

RTC_DCHECK_NE

[the sun, 2015/10/26 14:44:02]

Done.

+    stats.codec_name = codec_inst.plname;
+
+    // Get data from the last remote RTCP report.
+    std::vector<webrtc::ReportBlock> blocks;
+    if (rtp->GetRemoteRTCPReportBlocks(config_.voe_channel_id, &blocks) != -1) {
+      for (const webrtc::ReportBlock& block : blocks) {
+        // Lookup report for send ssrc only.
+        if (block.source_SSRC == stats.local_ssrc) {
+          stats.packets_lost = block.cumulative_num_packets_lost;
+          stats.fraction_lost = Q8ToFloat(block.fraction_lost);
+          stats.ext_seqnum = block.extended_highest_sequence_number;
+          // Convert samples to milliseconds.
+          if (codec_inst.plfreq / 1000 > 0) {
+            stats.jitter_ms =
+                block.interarrival_jitter / (codec_inst.plfreq / 1000);
+          }
+          break;
+        }
+      }
+    }
+  }
+
+  // RTT isn't known until a RTCP report is received. Until then, VoiceEngine
+  // returns 0 to indicate an error value.
+  if (call_stats.rttMs > 0) {
+    stats.rtt_ms = call_stats.rttMs;
+  }
+
+  // Local speech level.
+  {
+    unsigned int level = 0;
+    if (volume->GetSpeechInputLevelFullRange(level) != -1) {
+      stats.audio_level = static_cast<int32_t>(level);
+    }
+  }
+
+  // TODO(ajm): Re-enable this metric once we have a reliable implementation.
+  stats.aec_quality_min = -1;
+
+  bool echo_metrics_on = false;
+  if (processing->GetEcMetricsStatus(echo_metrics_on) != -1 &&
+      echo_metrics_on) {
+    // These can also be negative, but in practice -1 is only used to signal
+    // insufficient data, since the resolution is limited to multiples of 4 ms.

[kwiberg-webrtc, 2015/10/26 13:19:53]

:-)

[the sun, 2015/10/26 14:44:02]

This is actually copy+paste. There's some likelihood that we can use a better
method...

+    int median = -1;
+    int std = -1;
+    float dummy = 0.0f;
+    if (processing->GetEcDelayMetrics(median, std, dummy) != -1) {
+      stats.echo_delay_median_ms = median;
+      stats.echo_delay_std_ms = std;
+    }
+
+    // These can take on valid negative values, so use the lowest possible level
+    // as default rather than -1.

[kwiberg-webrtc, 2015/10/26 13:19:53]

:-)

[the sun, 2015/10/26 14:44:02]

Acknowledged.

+    int erl = -100;
+    int erle = -100;
+    int dummy1 = 0;
+    int dummy2 = 0;
+    if (processing->GetEchoMetrics(erl, erle, dummy1, dummy2) != -1) {
+      stats.echo_return_loss = erl;
+      stats.echo_return_loss_enhancement = erle;
+    }
+  }
+
+  // TODO(solenberg): Collect typing noise warnings here too!
+  // bool typing_noise_detected = typing_noise_detected_;
+
+  return stats;
+}
+
+const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return config_;
 }
 
 void AudioSendStream::Start() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
 void AudioSendStream::Stop() {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
 void AudioSendStream::SignalNetworkState(NetworkState state) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
 bool AudioSendStream::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;
 }
 }  // namespace internal
 }  // namespace webrtc