Bug 1780282 - Implement some more inbound rtp stats. r=ng,emilio

These are the inbound rtp stats that are P2 for MS Teams. Differential Revision: https://phabricator.services.mozilla.com/D152213
2022-07-22 00:47:31 +00:00 · 2022-07-22 00:47:31 +00:00 · 2a6bb59540
--- a/dom/media/webrtc/WebrtcGlobal.h
+++ b/dom/media/webrtc/WebrtcGlobal.h
@ -300,12 +300,17 @@ struct ParamTraits<mozilla::dom::RTCInboundRtpStreamStats> {
    WriteParam(aWriter, aParam.mFramesDecoded);
    WriteParam(aWriter, aParam.mFrameWidth);
    WriteParam(aWriter, aParam.mFrameHeight);
+    WriteParam(aWriter, aParam.mFramesPerSecond);
    WriteParam(aWriter, aParam.mBytesReceived);
    WriteParam(aWriter, aParam.mNackCount);
    WriteParam(aWriter, aParam.mFirCount);
    WriteParam(aWriter, aParam.mPliCount);
-    WriteParam(aWriter, aParam.mFramesPerSecond);
    WriteParam(aWriter, aParam.mFramesReceived);
+    WriteParam(aWriter, aParam.mJitterBufferDelay);
+    WriteParam(aWriter, aParam.mJitterBufferEmittedCount);
+    WriteParam(aWriter, aParam.mTotalSamplesReceived);
+    WriteParam(aWriter, aParam.mConcealedSamples);
+    WriteParam(aWriter, aParam.mSilentConcealedSamples);
    WriteRTCReceivedRtpStreamStats(aWriter, aParam);
  }

@ -314,12 +319,17 @@ struct ParamTraits<mozilla::dom::RTCInboundRtpStreamStats> {
           ReadParam(aReader, &(aResult->mFramesDecoded)) &&
           ReadParam(aReader, &(aResult->mFrameWidth)) &&
           ReadParam(aReader, &(aResult->mFrameHeight)) &&
+           ReadParam(aReader, &(aResult->mFramesPerSecond)) &&
           ReadParam(aReader, &(aResult->mBytesReceived)) &&
           ReadParam(aReader, &(aResult->mNackCount)) &&
           ReadParam(aReader, &(aResult->mFirCount)) &&
           ReadParam(aReader, &(aResult->mPliCount)) &&
-           ReadParam(aReader, &(aResult->mFramesPerSecond)) &&
           ReadParam(aReader, &(aResult->mFramesReceived)) &&
+           ReadParam(aReader, &(aResult->mJitterBufferDelay)) &&
+           ReadParam(aReader, &(aResult->mJitterBufferEmittedCount)) &&
+           ReadParam(aReader, &(aResult->mTotalSamplesReceived)) &&
+           ReadParam(aReader, &(aResult->mConcealedSamples)) &&
+           ReadParam(aReader, &(aResult->mSilentConcealedSamples)) &&
           ReadRTCReceivedRtpStreamStats(aReader, aResult);
  }
 };
--- a/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp
+++ b/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp
@ -314,6 +314,15 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal() {
              local.mPacketsReceived.Construct(audioStats->packets_rcvd);
              local.mPacketsDiscarded.Construct(audioStats->packets_discarded);
              local.mBytesReceived.Construct(audioStats->payload_bytes_rcvd);
+              local.mJitterBufferDelay.Construct(
+                  audioStats->jitter_buffer_delay_seconds);
+              local.mJitterBufferEmittedCount.Construct(
+                  audioStats->jitter_buffer_emitted_count);
+              local.mTotalSamplesReceived.Construct(
+                  audioStats->total_samples_received);
+              local.mConcealedSamples.Construct(audioStats->concealed_samples);
+              local.mSilentConcealedSamples.Construct(
+                  audioStats->silent_concealed_samples);
              /*
               * Potential new stats that are now available upstream.
              if (audioStats->last_packet_received_timestamp_ms) {
@ -335,15 +344,6 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal() {
                        webrtc::Timestamp::Millis(
                            *audioStats->estimated_playout_ntp_timestamp_ms)));
              }
-              local.mJitterBufferDelay.Construct(
-                  audioStats->jitter_buffer_delay_seconds);
-              local.mJitterBufferEmittedCount.Construct(
-                  audioStats->jitter_buffer_emitted_count);
-              local.mTotalSamplesReceived.Construct(
-                  audioStats->total_samples_received);
-              local.mConcealedSamples.Construct(audioStats->concealed_samples);
-              local.mSilentConcealedSamples.Construct(
-                  audioStats->silent_concealed_samples);
              local.mConcealmentEvents.Construct(
                  audioStats->concealment_events);
              local.mInsertedSamplesForDeceleration.Construct(
@ -435,7 +435,10 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal() {
              local.mFramesReceived.Construct(
                  videoStats->frame_counts.key_frames +
                  videoStats->frame_counts.delta_frames);
-
+              local.mJitterBufferDelay.Construct(
+                  videoStats->jitter_buffer_delay_seconds);
+              local.mJitterBufferEmittedCount.Construct(
+                  videoStats->jitter_buffer_emitted_count);
              /*
               * Potential new stats that are now available upstream.
              if (videoStats->qp_sum) {
@ -464,10 +467,6 @@ nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal() {
                        webrtc::Timestamp::Millis(
                            *videoStats->estimated_playout_ntp_timestamp_ms)));
              }
-              local.mJitterBufferDelay.Construct(
-                  videoStats->jitter_buffer_delay_seconds);
-              local.mJitterBufferEmittedCount.Construct(
-                  videoStats->jitter_buffer_emitted_count);
              local.mFramesReceived.Construct(
                  videoStats->frame_counts.key_frames +
                  videoStats->frame_counts.delta_frames);
--- a/dom/media/webrtc/tests/mochitests/stats.js
+++ b/dom/media/webrtc/tests/mochitests/stats.js
@ -19,6 +19,8 @@ const statsExpectedByType = {
      "packetsDiscarded",
      "bytesReceived",
      "jitter",
+      "jitterBufferDelay",
+      "jitterBufferEmittedCount",
    ],
    optional: ["remoteId", "nackCount"],
    localVideoOnly: [
@ -31,6 +33,11 @@ const statsExpectedByType = {
      "frameHeight",
      "framesReceived",
    ],
+    localAudioOnly: [
+      "totalSamplesReceived",
+      "concealedSamples",
+      "silentConcealedSamples",
+    ],
    unimplemented: [
      "mediaTrackId",
      "transportId",
@ -65,6 +72,7 @@ const statsExpectedByType = {
      "retransmittedBytesSent",
    ],
    optional: ["nackCount", "qpSum"],
+    localAudioOnly: [],
    localVideoOnly: [
      "framesEncoded",
      "firCount",
@ -230,7 +238,7 @@ const statsExpectedByType = {

 ["in", "out"].forEach(pre => {
  let s = statsExpectedByType[pre + "bound-rtp"];
-  s.optional = [...s.optional, ...s.localVideoOnly];
+  s.optional = [...s.optional, ...s.localVideoOnly, ...s.localAudioOnly];
 });

 //
@ -490,10 +498,67 @@ function pedanticChecks(report) {
          `local only test. value=${stat.jitter}`
      );

+      // jitterBufferEmittedCount
+      let expectedJitterBufferEmmitedCount = stat.kind == "video" ? 10 : 1000;
+      ok(
+        stat.jitterBufferEmittedCount > expectedJitterBufferEmmitedCount,
+        `${stat.type}.jitterBufferEmittedCount is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.jitterBufferEmittedCount}`
+      );
+
+      // jitterBufferDelay
+      let avgJitterBufferDelay =
+        stat.jitterBufferDelay / stat.jitterBufferEmittedCount;
+      ok(
+        avgJitterBufferDelay > 0.01 && avgJitterBufferDelay < 10,
+        `${stat.type}.jitterBufferDelay is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.jitterBufferDelay}/${stat.jitterBufferEmittedCount}=${avgJitterBufferDelay}`
+      );
+
      //
      // Optional fields
      //

+      //
+      // Local audio only stats
+      //
+      if (stat.inner.kind != "audio") {
+        expectations.localAudioOnly.forEach(field => {
+          ok(
+            stat[field] === undefined,
+            `${stat.type} does not have field ${field}` +
+              ` when kind is not 'audio'`
+          );
+        });
+      } else {
+        expectations.localAudioOnly.forEach(field => {
+          ok(
+            stat.inner[field] !== undefined,
+            stat.type + " has field " + field + " when kind is video"
+          );
+        });
+        // totalSamplesReceived
+        ok(
+          stat.totalSamplesReceived > 1000,
+          `${stat.type}.totalSamplesReceived is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.totalSamplesReceived}`
+        );
+
+        // concealedSamples
+        ok(
+          stat.concealedSamples > 100,
+          `${stat.type}.concealedSamples is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.concealedSamples}`
+        );
+
+        // silentConcealedSamples
+        ok(
+          stat.silentConcealedSamples >= 0,
+          `${stat.type}.silentConcealedSamples is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.silentConcealedSamples}`
+        );
+      }
+
      //
      // Local video only stats
      //
--- a/dom/webidl/RTCStatsReport.webidl
+++ b/dom/webidl/RTCStatsReport.webidl
@ -6,6 +6,7 @@
 * The origin of this IDL file is
 * http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcstatsreport-object
 * http://www.w3.org/2011/04/webrtc/wiki/Stats
+ * https://www.w3.org/TR/webrtc-stats/
 */

 enum RTCStatsType {
@ -66,11 +67,16 @@ dictionary RTCInboundRtpStreamStats : RTCReceivedRtpStreamStats {
  unsigned long framesDecoded;
  unsigned long frameWidth;
  unsigned long frameHeight;
+  double framesPerSecond;
  unsigned long long bytesReceived;
  unsigned long nackCount;
  unsigned long firCount;
  unsigned long pliCount;
-  double framesPerSecond;
+  double jitterBufferDelay;
+  unsigned long long jitterBufferEmittedCount;
+  unsigned long long totalSamplesReceived;
+  unsigned long long concealedSamples;
+  unsigned long long silentConcealedSamples;
  unsigned long framesReceived;
 };

--- a/testing/web-platform/meta/webrtc-stats/supported-stats.html.ini
+++ b/testing/web-platform/meta/webrtc-stats/supported-stats.html.ini
@ -35,21 +35,6 @@
  [inbound-rtp's estimatedPlayoutTimestamp]
    expected: FAIL

-  [inbound-rtp's jitterBufferDelay]
-    expected: FAIL
-
-  [inbound-rtp's jitterBufferEmittedCount]
-    expected: FAIL
-
-  [inbound-rtp's totalSamplesReceived]
-    expected: FAIL
-
-  [inbound-rtp's concealedSamples]
-    expected: FAIL
-
-  [inbound-rtp's silentConcealedSamples]
-    expected: FAIL
-
  [inbound-rtp's concealmentEvents]
    expected: FAIL