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 2607 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);
+    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 191 matching lines @@
       LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
       return false;
     }
   }
   return true;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE