Implement AudioReceiveStream::GetStats().

talk/media/webrtc/webrtcvoiceengine.cc

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 2676 matching lines @@
 
     int median, std;
     float dummy;
     if (engine()->voe()->processing()->GetEcDelayMetrics(
         median, std, dummy) != -1) {
       echo_delay_median_ms = median;
       echo_delay_std_ms = std;
     }
   }
 
-  webrtc::CallStatistics cs;
-  unsigned int ssrc;
-  webrtc::CodecInst codec;
-  unsigned int level;
-
   for (const auto& ch : send_channels_) {
     const int channel = ch.second->channel();
 
     // Fill in the sender info, based on what we know, and what the
     // remote side told us it got from its RTCP report.
     VoiceSenderInfo sinfo;
 
+    webrtc::CallStatistics cs = {0};
+    unsigned int ssrc = 0;
     if (engine()->voe()->rtp()->GetRTCPStatistics(channel, cs) == -1 ||
         engine()->voe()->rtp()->GetLocalSSRC(channel, ssrc) == -1) {
       continue;
     }
 
     sinfo.add_ssrc(ssrc);
     sinfo.codec_name = send_codec_.get() ? send_codec_->plname : "";
     sinfo.bytes_sent = cs.bytesSent;
     sinfo.packets_sent = cs.packetsSent;
     // RTT isn't known until a RTCP report is received. Until then, VoiceEngine
     // returns 0 to indicate an error value.
     sinfo.rtt_ms = (cs.rttMs > 0) ? cs.rttMs : -1;
 
     // Get data from the last remote RTCP report. Use default values if no data
     // available.
     sinfo.fraction_lost = -1.0;
     sinfo.jitter_ms = -1;
     sinfo.packets_lost = -1;
     sinfo.ext_seqnum = -1;
     std::vector<webrtc::ReportBlock> receive_blocks;
+    webrtc::CodecInst codec = {0};
     if (engine()->voe()->rtp()->GetRemoteRTCPReportBlocks(
             channel, &receive_blocks) != -1 &&
         engine()->voe()->codec()->GetSendCodec(channel, codec) != -1) {
       for (const webrtc::ReportBlock& block : receive_blocks) {
         // Lookup report for send ssrc only.
         if (block.source_SSRC == sinfo.ssrc()) {
           // Convert Q8 to floating point.
           sinfo.fraction_lost = static_cast<float>(block.fraction_lost) / 256;
           // Convert samples to milliseconds.
           if (codec.plfreq / 1000 > 0) {
             sinfo.jitter_ms = block.interarrival_jitter / (codec.plfreq / 1000);
           }
           sinfo.packets_lost = block.cumulative_num_packets_lost;
           sinfo.ext_seqnum = block.extended_highest_sequence_number;
           break;
         }
       }
     }
 
     // Local speech level.
+    unsigned int level = 0;
     sinfo.audio_level = (engine()->voe()->volume()->
         GetSpeechInputLevelFullRange(level) != -1) ? level : -1;
 
     // TODO(xians): We are injecting the same APM logging to all the send
     // channels here because there is no good way to know which send channel
     // is using the APM. The correct fix is to allow the send channels to have
     // their own APM so that we can feed the correct APM logging to different
     // send channels. See issue crbug/264611 .
     sinfo.echo_return_loss = echo_return_loss;
     sinfo.echo_return_loss_enhancement = echo_return_loss_enhancement;
     sinfo.echo_delay_median_ms = echo_delay_median_ms;
     sinfo.echo_delay_std_ms = echo_delay_std_ms;
     // TODO(ajm): Re-enable this metric once we have a reliable implementation.
     sinfo.aec_quality_min = -1;
     sinfo.typing_noise_detected = typing_noise_detected_;
 
     info->senders.push_back(sinfo);
   }
 
   // Get the SSRC and stats for each receiver.
-  for (const auto& ch : receive_channels_) {
-    int ch_id = ch.second->channel();
-    memset(&cs, 0, sizeof(cs));
-    if (engine()->voe()->rtp()->GetRemoteSSRC(ch_id, ssrc) != -1 &&
-        engine()->voe()->rtp()->GetRTCPStatistics(ch_id, cs) != -1 &&
-        engine()->voe()->codec()->GetRecCodec(ch_id, codec) != -1) {
-      VoiceReceiverInfo rinfo;
-      rinfo.add_ssrc(ssrc);
-      rinfo.bytes_rcvd = cs.bytesReceived;
-      rinfo.packets_rcvd = cs.packetsReceived;
-      // The next four fields are from the most recently sent RTCP report.
-      // Convert Q8 to floating point.
-      rinfo.fraction_lost = static_cast<float>(cs.fractionLost) / (1 << 8);
-      rinfo.packets_lost = cs.cumulativeLost;
-      rinfo.ext_seqnum = cs.extendedMax;
-      rinfo.capture_start_ntp_time_ms = cs.capture_start_ntp_time_ms_;
-      if (codec.pltype != -1) {
-        rinfo.codec_name = codec.plname;
-      }
-      // Convert samples to milliseconds.
-      if (codec.plfreq / 1000 > 0) {
-        rinfo.jitter_ms = cs.jitterSamples / (codec.plfreq / 1000);
-      }
-
-      // Get jitter buffer and total delay (alg + jitter + playout) stats.
-      webrtc::NetworkStatistics ns;
-      if (engine()->voe()->neteq() &&
-          engine()->voe()->neteq()->GetNetworkStatistics(
-              ch_id, ns) != -1) {
-        rinfo.jitter_buffer_ms = ns.currentBufferSize;
-        rinfo.jitter_buffer_preferred_ms = ns.preferredBufferSize;
-        rinfo.expand_rate =
-            static_cast<float>(ns.currentExpandRate) / (1 << 14);
-        rinfo.speech_expand_rate =
-            static_cast<float>(ns.currentSpeechExpandRate) / (1 << 14);
-        rinfo.secondary_decoded_rate =
-            static_cast<float>(ns.currentSecondaryDecodedRate) / (1 << 14);
-        rinfo.accelerate_rate =
-            static_cast<float>(ns.currentAccelerateRate) / (1 << 14);
-        rinfo.preemptive_expand_rate =
-            static_cast<float>(ns.currentPreemptiveRate) / (1 << 14);
-      }
-
-      webrtc::AudioDecodingCallStats ds;
-      if (engine()->voe()->neteq() &&
-          engine()->voe()->neteq()->GetDecodingCallStatistics(
-              ch_id, &ds) != -1) {
-        rinfo.decoding_calls_to_silence_generator =
-            ds.calls_to_silence_generator;
-        rinfo.decoding_calls_to_neteq = ds.calls_to_neteq;
-        rinfo.decoding_normal = ds.decoded_normal;
-        rinfo.decoding_plc = ds.decoded_plc;
-        rinfo.decoding_cng = ds.decoded_cng;
-        rinfo.decoding_plc_cng = ds.decoded_plc_cng;
-      }
-
-      if (engine()->voe()->sync()) {
-        int jitter_buffer_delay_ms = 0;
-        int playout_buffer_delay_ms = 0;
-        engine()->voe()->sync()->GetDelayEstimate(
-            ch_id, &jitter_buffer_delay_ms, &playout_buffer_delay_ms);
-        rinfo.delay_estimate_ms = jitter_buffer_delay_ms +
-            playout_buffer_delay_ms;
-      }
-
-      // Get speech level.
-      rinfo.audio_level = (engine()->voe()->volume()->
-          GetSpeechOutputLevelFullRange(ch_id, level) != -1) ? level : -1;
-      info->receivers.push_back(rinfo);
-    }
+  info->receivers.clear();
+  for (const auto& stream : receive_streams_) {
+    webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats();
+    VoiceReceiverInfo rinfo;
+    rinfo.add_ssrc(stats.remote_ssrc);

[tommi, 2015/10/19 12:36:24]

What do you think about moving all this boiler plate to a VoiceReceiveInfo ctor?

[the sun, 2015/10/19 14:25:01]

You mean forcing VoiceReceiveInfo to know about the AudioReceiveStream::Stats? I
think that is breaking encapsulation. We'll be better off deprecating
VoiceReceiverInfo, going forward.

[tommi, 2015/10/19 14:55:44]

OK, sgtm. There are a lot of vars being set here so my thinking was that if we
need to do this elsewhere, it would be good to have one place where all this is
and possibly have a ctor for it too so that if one forgets one variable, things
don't compile.  If this is a one-off thing though, then fine with me as is.

[the sun, 2015/10/20 08:31:29]

Acknowledged.

+    rinfo.bytes_rcvd = stats.bytes_rcvd;
+    rinfo.packets_rcvd = stats.packets_rcvd;
+    rinfo.packets_lost = stats.packets_lost;
+    rinfo.fraction_lost = stats.fraction_lost;
+    rinfo.codec_name = stats.codec_name;
+    rinfo.ext_seqnum = stats.ext_seqnum;
+    rinfo.jitter_ms = stats.jitter_ms;
+    rinfo.jitter_buffer_ms = stats.jitter_buffer_ms;
+    rinfo.jitter_buffer_preferred_ms = stats.jitter_buffer_preferred_ms;
+    rinfo.delay_estimate_ms = stats.delay_estimate_ms;
+    rinfo.audio_level = stats.audio_level;
+    rinfo.expand_rate = stats.expand_rate;
+    rinfo.speech_expand_rate = stats.speech_expand_rate;
+    rinfo.secondary_decoded_rate = stats.secondary_decoded_rate;
+    rinfo.accelerate_rate = stats.accelerate_rate;
+    rinfo.preemptive_expand_rate = stats.preemptive_expand_rate;
+    rinfo.decoding_calls_to_silence_generator =
+        stats.decoding_calls_to_silence_generator;
+    rinfo.decoding_calls_to_neteq = stats.decoding_calls_to_neteq;
+    rinfo.decoding_normal = stats.decoding_normal;
+    rinfo.decoding_plc = stats.decoding_plc;
+    rinfo.decoding_cng = stats.decoding_cng;
+    rinfo.decoding_plc_cng = stats.decoding_plc_cng;
+    rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms;
+    info->receivers.push_back(rinfo);
   }
 
   return true;
 }
 
 void WebRtcVoiceMediaChannel::OnError(int error) {
   if (send_ == SEND_NOTHING) {
     return;
   }
   if (error == VE_TYPING_NOISE_WARNING) {
@@ skipping 203 matching lines @@
       LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
       return false;
     }
   }
   return true;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE