Implement AudioReceiveStream::GetStats().

talk/media/webrtc/webrtcvoiceengine.cc

Index: talk/media/webrtc/webrtcvoiceengine.cc
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 2a3df2db635c3a815a668865950513c24217fdbb..d880e4bdcabb3fcb4a28097e6f0520a566cf8b78 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -2694,11 +2694,6 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
     }
   }
 
-  webrtc::CallStatistics cs;
-  unsigned int ssrc;
-  webrtc::CodecInst codec;
-  unsigned int level;
-
   for (const auto& ch : send_channels_) {
     const int channel = ch.second->channel();
 
@@ -2706,6 +2701,8 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
     // 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;
@@ -2726,6 +2723,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
     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) {
@@ -2746,6 +2744,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
     }
 
     // Local speech level.
+    unsigned int level = 0;
     sinfo.audio_level = (engine()->voe()->volume()->
         GetSpeechInputLevelFullRange(level) != -1) ? level : -1;
 
@@ -2766,76 +2765,36 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
   }
 
   // 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;