Android WebRTC - getStats() is not providing enough information

Question

I am trying to get all information related to mediastreams to get call quality. Peerconnection.getStats() method is deprecated but provides all information as per my requirement like "bytesReceived", "packetsLost", "packetsReceived" ,"googCodecName" and "googJitterBufferMs".

peerConnection.getStats(reports -> {
   for (StatsReport report : reports) {
      Log.d(TAG, "Stats: " + report.toString());
   }
}, null);

Response:

{
values: [
    bytesReceived: 951618
  ],
  [
    codecImplementationName: OMX.qcom.video.decoder.vp8
  ],
  [
    framesDecoded: 171
  ],
  [
    mediaType: video
  ],
  [
    packetsLost: 4
  ],
  [
    packetsReceived: 908
  ],
  [
    qpSum: 6409
  ],
  [
    ssrc: 3637617127
  ],
  [
    transportId: Channel-audio-1
  ],
  [
    googCaptureStartNtpTimeMs: 3766113175824
  ],
  [
    googCodecName: VP8
  ],
  [
    googContentType: realtime
  ],
  [
    googCurrentDelayMs: 196
  ],
  [
    googDecodeMs: 54
  ],
  [
    googFirsSent: 0
  ],
  [
    googFirstFrameReceivedToDecodedMs: 225
  ],
  [
    googFrameHeightReceived: 720
  ],
  [
    googFrameRateDecoded: 22
  ],
  [
    googFrameRateOutput: 22
  ],
  [
    googFrameRateReceived: 20
  ],
  [
    googFrameWidthReceived: 960
  ],
  [
    googInterframeDelayMax: 55
  ],
  [
    googJitterBufferMs: 116
  ],
  [
    googMaxDecodeMs: 70
  ],
  [
    googMinPlayoutDelayMs: 91
  ],
  [
    googNacksSent: 6
  ],
  [
    googPlisSent: 0
  ],
  [
    googRenderDelayMs: 10
  ],
  [
    googTargetDelayMs: 196
  ],
  [
    googTrackId: 100
  ]
}

Now I can't use this method as its deprecated. When I am trying to use new getStats() method, It does not provides all such information and also the response is also very unorganised.

peerConnection.getStats(new RTCStatsCollectorCallback() {
          @Override
          public void onStatsDelivered(RTCStatsReport rtcStatsReport) {
                 Log.d(TAG, "RTCStatsReport: "+rtcStatsReport.getStatsMap().toString());
          }
});

Response:-

    {
  RTCCodec_audio_Inbound_0={
    timestampUs: 1557124386437087,
    type: codec,
    id: RTCCodec_audio_Inbound_0,
    payloadType: 0,
    mimeType: "audio/PCMU",
    clockRate: 8000
  },
  RTCCodec_audio_Inbound_102={
    timestampUs: 1557124386437087,
    type: codec,
    id: RTCCodec_audio_Inbound_102,
    payloadType: 102,
    mimeType: "audio/ILBC",
    clockRate: 8000
  },
  RTCCodec_audio_Inbound_103={
    timestampUs: 1557124386437087,
    type: codec,
    id: RTCCodec_audio_Inbound_103,
    payloadType: 103,
    mimeType: "audio/ISAC",
    clockRate: 16000
  },
  RTCCodec_audio_Inbound_105={
    timestampUs: 1557124386437087,
    type: codec,
    id: RTCCodec_audio_Inbound_105,
    payloadType: 105,
    mimeType: "audio/CN",
    clockRate: 16000
  },
  RTCCodec_audio_Inbound_110={
    timestampUs: 1557124386437087,
    type: codec,
    id: RTCCodec_audio_Inbound_110,
    payloadType: 110,
    mimeType: "audio/telephone-event",
    clockRate: 48000
  },
  RTCCodec_audio_Inbound_111={
    timestampUs: 1557124386437087,
    type: codec,
    id: RTCCodec_audio_Inbound_111,
    payloadType: 111,
    mimeType: "audio/opus",
    clockRate: 48000
  },
  RTCCodec_audio_Inbound_113={
    timestampUs: 1557124386437087,
    type: codec,
    id: RTCCodec_audio_Inbound_113,
    payloadType: 113,
    mimeType: "audio/telephone-event",
    clockRate: 16000
  },
  RTCCodec_audio_Inbound_126={
    timestampUs: 1557124386437087,
    type: codec,
    id: RTCCodec_audio_Inbound_126,
    payloadType: 126,
    mimeType: "audio/telephone-event",
    clockRate: 8000
  },
  RTCCodec_audio_Inbound_13={
    timestampUs: 1557124386437087,
    type: codec,
    id: RTCCodec_audio_Inbound_13,
    payloadType: 13,
    mimeType: "audio/CN",
    clockRate: 8000
  },
  RTCCodec_audio_Inbound_8={
    timestampUs: 1557124386437087,
    type: codec,
    id: RTCCodec_audio_Inbound_8,
    payloadType: 8,
    mimeType: "audio/PCMA",
    clockRate: 8000
  },
  RTCCodec_audio_Inbound_9={
    timestampUs: 1557124386437087,
    type: codec,
    id: RTCCodec_audio_Inbound_9,
    payloadType: 9,
    mimeType: "audio/G722",
    clockRate: 8000
  },
  RTCCodec_audio_Outbound_0={
    timestampUs: 1557124386437087,
    type: codec,
    id: RTCCodec_audio_Outbound_0,
    payloadType: 0,
    mimeType: "audio/PCMU",
    clockRate: 8000
  },
  RTCCodec_audio_Outbound_102={
    timestampUs: 1557124386437087,
    type: codec,
    id: RTCCodec_audio_Outbound_102,
    payloadType: 102,
    mimeType: "audio/ILBC",
    clockRate: 8000
  },
  RTCCodec_audio_Outbound_103={
    timestampUs: 1557124386437087,
    type: codec,
    id: RTCCodec_audio_Outbound_103,
    payloadType: 103,
    mimeType: "audio/ISAC",
    clockRate: 16000
  }
}

I am hitting getStats() method after every second and each time it gives me response with different data. This response is not documented anywhere on WebRTC docs.

How I can get "bytesReceived", "packetsLost", "packetsReceived" ,"googCodecName" and "googJitterBufferMs" with new getStats() method.

@Nitin Gupta I am also facing the same issue. Have you found any solution? — abhishek kumar gupta, Jul 06 '20 at 12:03

Paul Oliva · Answer 1 · 2020-12-02T21:05:05.827

I see this on mine:

{ timestampUs: 1606895449567493, type: inbound-rtp, id: RTCInboundRTPVideoStream_512, ssrc: 512, isRemote: false, mediaType: "video", kind: "video", trackId: "RTCMediaStreamTrack_receiver_2", transportId: "RTCTransport_1_1", firCount: 0, pliCount: 0, nackCount: 0, packetsReceived: 0, bytesReceived: 0, headerBytesReceived: 0, packetsLost: 0, framesReceived: 0, framesDecoded: 0, keyFramesDecoded: 0, framesDropped: 0, totalDecodeTime: 0.0, totalInterFrameDelay: 0.0, totalSquaredInterFrameDelay: 0.0, decoderImplementation: "unknown" },
{ timestampUs: 1606895449567493, type: track, id: RTCMediaStreamTrack_receiver_1, trackIdentifier: "c68ef0fb-7ac9-4008-97db-100f9a04c66e", remoteSource: true, ended: false, detached: false, kind: "audio", jitterBufferDelay: 0.0, jitterBufferEmittedCount: 0, totalAudioEnergy: 0.0, totalSamplesReceived: 0, totalSamplesDuration: 0.0, concealedSamples: 0, silentConcealedSamples: 0, concealmentEvents: 0, insertedSamplesForDeceleration: 0, removedSamplesForAcceleration: 0, jitterBufferFlushes: 0, delayedPacketOutageSamples: 0, relativePacketArrivalDelay: 0.0, jitterBufferTargetDelay: 0.0, interruptionCount: 0, totalInterruptionDuration: 0.0 },
{ timestampUs: 1606895449567493, type: track, id: RTCMediaStreamTrack_receiver_2, trackIdentifier: "05c66233-cb7f-4984-9d27-3dde1ef92a36", remoteSource: true, ended: false, detached: false, kind: "video", jitterBufferDelay: 0.0, jitterBufferEmittedCount: 0, framesReceived: 0, framesDecoded: 0, framesDropped: 0, freezeCount: 0, pauseCount: 0, totalFreezesDuration: 0.0, totalPausesDuration: 0.0, totalFramesDuration: 0.0, sumOfSquaredFramesDuration: 0.0 }

Can you look for those in your stats?

I call it like this:

peerConnection.getStats(new RTCStatsCollectorCallback() {
  @Override
  public void onStatsDelivered(RTCStatsReport rtcStatsReport) {
    longInfo("RTC Stats: \n" + rtcStatsReport.toString());
  }
});

Edit: Since the stats report is so long, you will need to make a longInfo method like this to see everything. That should solve your problem.

  public void longInfo(String str) {
    if (str.length() > 4000) {
        Log.i(TAG, str.substring(0, 4000));
        longInfo(str.substring(4000));
    } else
        Log.i(TAG, str);
    }

Android WebRTC - getStats() is not providing enough information

1 Answers1