Implement AudioReceiveStream::GetStats().

talk/media/webrtc/fakewebrtcvoiceengine.h

Index: talk/media/webrtc/fakewebrtcvoiceengine.h
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index 1167b6b9830c2f9edd1ff0b9caa606a50814d372..9b913276a6231aa4c1fab423dfeffdbb62b8b5cd 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -65,25 +65,6 @@ static const int kOpusBandwidthWb = 8000;
 static const int kOpusBandwidthSwb = 12000;
 static const int kOpusBandwidthFb = 20000;
 
-static const webrtc::NetworkStatistics kNetStats = {
-    1,  // uint16_t currentBufferSize;
-    2,  // uint16_t preferredBufferSize;
-    true,  // bool jitterPeaksFound;
-    1234,  // uint16_t currentPacketLossRate;
-    567,   // uint16_t currentDiscardRate;
-    8901,  // uint16_t currentExpandRate;
-    234,  // uint16_t currentSpeechExpandRate;
-    5678, // uint16_t currentPreemptiveRate;
-    9012, // uint16_t currentAccelerateRate;
-    3456, // uint16_t currentSecondaryDecodedRate;
-    7890, // int32_t clockDriftPPM;
-    54,  // meanWaitingTimeMs;
-    32,  // int medianWaitingTimeMs;
-    1,  // int minWaitingTimeMs;
-    98, // int maxWaitingTimeMs;
-    7654,  // int addedSamples;
-};  // These random but non-trivial numbers are used for testing.
-
 #define WEBRTC_CHECK_CHANNEL(channel) \
   if (channels_.find(channel) == channels_.end()) return -1;
 
@@ -181,9 +162,9 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
 class FakeWebRtcVoiceEngine
     : public webrtc::VoEAudioProcessing,
       public webrtc::VoEBase, public webrtc::VoECodec, public webrtc::VoEDtmf,
-      public webrtc::VoEHardware, public webrtc::VoENetEqStats,
+      public webrtc::VoEHardware,
       public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
-      public webrtc::VoEVideoSync, public webrtc::VoEVolumeControl {
+      public webrtc::VoEVolumeControl {
  public:
   struct DtmfInfo {
     DtmfInfo()
@@ -527,26 +508,7 @@ class FakeWebRtcVoiceEngine
     return 0;
   }
   WEBRTC_STUB(SetBitRate, (int channel, int bitrate_bps));
-  WEBRTC_FUNC(GetRecCodec, (int channel, webrtc::CodecInst& codec)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    const Channel* c = channels_[channel];
-    for (std::list<std::string>::const_iterator it_packet = c->packets.begin();
-        it_packet != c->packets.end(); ++it_packet) {
-      int pltype;
-      if (!GetRtpPayloadType(it_packet->data(), it_packet->length(), &pltype)) {
-        continue;
-      }
-      for (std::vector<webrtc::CodecInst>::const_iterator it_codec =
-          c->recv_codecs.begin(); it_codec != c->recv_codecs.end();
-          ++it_codec) {
-        if (it_codec->pltype == pltype) {
-          codec = *it_codec;
-          return 0;
-        }
-      }
-    }
-    return -1;
-  }
+  WEBRTC_STUB(GetRecCodec, (int channel, webrtc::CodecInst& codec));
   WEBRTC_FUNC(SetRecPayloadType, (int channel,
                                   const webrtc::CodecInst& codec)) {
     WEBRTC_CHECK_CHANNEL(channel);
@@ -725,20 +687,6 @@ class FakeWebRtcVoiceEngine
   WEBRTC_STUB(EnableBuiltInNS, (bool enable));
   virtual bool BuiltInNSIsAvailable() const { return false; }
 
-  // webrtc::VoENetEqStats
-  WEBRTC_FUNC(GetNetworkStatistics, (int channel,
-                                     webrtc::NetworkStatistics& ns)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    memcpy(&ns, &kNetStats, sizeof(webrtc::NetworkStatistics));
-    return 0;
-  }
-
-  WEBRTC_FUNC_CONST(GetDecodingCallStatistics, (int channel,
-      webrtc::AudioDecodingCallStats*)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    return 0;
-  }
-
   // webrtc::VoENetwork
   WEBRTC_FUNC(RegisterExternalTransport, (int channel,
                                           webrtc::Transport& transport)) {
@@ -887,18 +835,6 @@ class FakeWebRtcVoiceEngine
     return 0;
   }
 
-  // webrtc::VoEVideoSync
-  WEBRTC_STUB(GetPlayoutBufferSize, (int& bufferMs));
-  WEBRTC_STUB(GetPlayoutTimestamp, (int channel, unsigned int& timestamp));
-  WEBRTC_STUB(GetRtpRtcp, (int, webrtc::RtpRtcp**, webrtc::RtpReceiver**));
-  WEBRTC_STUB(SetInitTimestamp, (int channel, unsigned int timestamp));
-  WEBRTC_STUB(SetInitSequenceNumber, (int channel, short sequenceNumber));
-  WEBRTC_STUB(SetMinimumPlayoutDelay, (int channel, int delayMs));
-  WEBRTC_STUB(SetInitialPlayoutDelay, (int channel, int delay_ms));
-  WEBRTC_STUB(GetDelayEstimate, (int channel, int* jitter_buffer_delay_ms,
-                                 int* playout_buffer_delay_ms));
-  WEBRTC_STUB_CONST(GetLeastRequiredDelayMs, (int channel));
-
   // webrtc::VoEVolumeControl
   WEBRTC_STUB(SetSpeakerVolume, (unsigned int));
   WEBRTC_STUB(GetSpeakerVolume, (unsigned int&));