diff --git a/resources/audio_coding/neteq4_network_stats_android.dat.sha1 b/resources/audio_coding/neteq4_network_stats_android.dat.sha1
new file mode 100644
index 0000000000..56907e15ee
--- /dev/null
+++ b/resources/audio_coding/neteq4_network_stats_android.dat.sha1
@@ -0,0 +1 @@
+2853ab577fe571adfc7b18f77bbe58f1253d2019
\ No newline at end of file
diff --git a/resources/audio_coding/neteq4_rtcp_stats_android.dat.sha1 b/resources/audio_coding/neteq4_rtcp_stats_android.dat.sha1
new file mode 100644
index 0000000000..558d42dc16
--- /dev/null
+++ b/resources/audio_coding/neteq4_rtcp_stats_android.dat.sha1
@@ -0,0 +1 @@
+f3f7b3d3e71d7e635240b5373b57df6a7e4ce9d4
\ No newline at end of file
diff --git a/resources/audio_coding/neteq4_universal_ref_android.pcm.sha1 b/resources/audio_coding/neteq4_universal_ref_android.pcm.sha1
new file mode 100644
index 0000000000..4243b2aad5
--- /dev/null
+++ b/resources/audio_coding/neteq4_universal_ref_android.pcm.sha1
@@ -0,0 +1 @@
+133f5c18838c1d5c3cc61744e1e5f4d144518239
\ No newline at end of file
diff --git a/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
index d3f59ec668..0b4754d970 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
@@ -161,7 +161,7 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
     return next_send_time;
   }
 
-  void VerifyOutput(size_t num_samples) {
+  virtual void VerifyOutput(size_t num_samples) {
     for (size_t i = 0; i < num_samples; ++i) {
       for (size_t j = 0; j < num_channels_; ++j) {
         ASSERT_EQ(output_[i], output_multi_channel_[i * num_channels_ + j]) <<
@@ -275,12 +275,7 @@ class NetEqStereoTestNoJitter : public NetEqStereoTest {
   }
 };
 
-#if defined(WEBRTC_ANDROID)
-#define MAYBE_RunTest DISABLED_RunTest
-#else
-#define MAYBE_RunTest RunTest
-#endif
-TEST_P(NetEqStereoTestNoJitter, MAYBE_RunTest) {
+TEST_P(NetEqStereoTestNoJitter, RunTest) {
   RunTest(8);
 }
 
@@ -305,7 +300,7 @@ class NetEqStereoTestPositiveDrift : public NetEqStereoTest {
   double drift_factor;
 };
 
-TEST_P(NetEqStereoTestPositiveDrift, MAYBE_RunTest) {
+TEST_P(NetEqStereoTestPositiveDrift, RunTest) {
   RunTest(100);
 }
 
@@ -318,7 +313,7 @@ class NetEqStereoTestNegativeDrift : public NetEqStereoTestPositiveDrift {
   }
 };
 
-TEST_P(NetEqStereoTestNegativeDrift, MAYBE_RunTest) {
+TEST_P(NetEqStereoTestNegativeDrift, RunTest) {
   RunTest(100);
 }
 
@@ -346,7 +341,7 @@ class NetEqStereoTestDelays : public NetEqStereoTest {
   int frame_index_;
 };
 
-TEST_P(NetEqStereoTestDelays, MAYBE_RunTest) {
+TEST_P(NetEqStereoTestDelays, RunTest) {
   RunTest(1000);
 }
 
@@ -362,13 +357,25 @@ class NetEqStereoTestLosses : public NetEqStereoTest {
     return (++frame_index_) % kLossInterval == 0;
   }
 
+  // TODO(hlundin): NetEq is not giving bitexact results for these cases.
+  virtual void VerifyOutput(size_t num_samples) {
+    for (size_t i = 0; i < num_samples; ++i) {
+      auto first_channel_sample = output_multi_channel_[i * num_channels_];
+      for (size_t j = 0; j < num_channels_; ++j) {
+        const int kErrorMargin = 200;
+        EXPECT_NEAR(output_[i], output_multi_channel_[i * num_channels_ + j],
+                    kErrorMargin)
+            << "Diff in sample " << i << ", channel " << j << ".";
+        EXPECT_EQ(first_channel_sample,
+                  output_multi_channel_[i * num_channels_ + j]);
+      }
+    }
+  }
+
   int frame_index_;
 };
 
-// TODO(pbos): Enable on non-Android, this went failing while being accidentally
-// disabled on all platforms and not just Android.
-// https://bugs.chromium.org/p/webrtc/issues/detail?id=5387
-TEST_P(NetEqStereoTestLosses, DISABLED_RunTest) {
+TEST_P(NetEqStereoTestLosses, RunTest) {
   RunTest(100);
 }
 
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index 8d52c615da..f218f721ae 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -412,6 +412,10 @@ void NetEqDecodingTest::Process(size_t* out_len) {
     if (packet_->payload_length_bytes() > 0) {
       WebRtcRTPHeader rtp_header;
       packet_->ConvertHeader(&rtp_header);
+#ifndef WEBRTC_CODEC_ISAC
+      // Ignore payload type 104 (iSAC-swb) if ISAC is not supported.
+      if (rtp_header.header.payloadType != 104)
+#endif
       ASSERT_EQ(0, neteq_->InsertPacket(
                        rtp_header,
                        rtc::ArrayView<const uint8_t>(
@@ -515,10 +519,10 @@ void NetEqDecodingTest::PopulateCng(int frame_index,
   *payload_len = 1;  // Only noise level, no spectral parameters.
 }
 
-#if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) &&             \
-    defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
-    (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
-    defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
+#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
+    (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) &&    \
+    defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722) &&        \
+    !defined(WEBRTC_ARCH_ARM64)
 #define MAYBE_TestBitExactness TestBitExactness
 #else
 #define MAYBE_TestBitExactness DISABLED_TestBitExactness
@@ -929,12 +933,12 @@ TEST_F(NetEqDecodingTest, UnknownPayloadType) {
   EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
 }
 
-#if defined(WEBRTC_ANDROID)
-#define MAYBE_DecoderError DISABLED_DecoderError
-#else
-#define MAYBE_DecoderError DecoderError
-#endif
 #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+#define MAYBE_DecoderError DecoderError
+#else
+#define MAYBE_DecoderError DISABLED_DecoderError
+#endif
+
 TEST_F(NetEqDecodingTest, MAYBE_DecoderError) {
   const size_t kPayloadBytes = 100;
   uint8_t payload[kPayloadBytes] = {0};
@@ -955,8 +959,16 @@ TEST_F(NetEqDecodingTest, MAYBE_DecoderError) {
                              &samples_per_channel, &num_channels, &type));
   // Verify that there is a decoder error to check.
   EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
-  // Code 6730 is an iSAC error code.
-  EXPECT_EQ(6730, neteq_->LastDecoderError());
+
+  enum NetEqDecoderError {
+    ISAC_LENGTH_MISMATCH = 6730,
+    ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH = 6640
+  };
+#if defined(WEBRTC_CODEC_ISAC)
+  EXPECT_EQ(ISAC_LENGTH_MISMATCH, neteq_->LastDecoderError());
+#elif defined(WEBRTC_CODEC_ISACFX)
+  EXPECT_EQ(ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH, neteq_->LastDecoderError());
+#endif
   // Verify that the first 160 samples are set to 0, and that the remaining
   // samples are left unmodified.
   static const int kExpectedOutputLength = 160;  // 10 ms at 16 kHz sample rate.
@@ -973,7 +985,6 @@ TEST_F(NetEqDecodingTest, MAYBE_DecoderError) {
     EXPECT_EQ(1, out_data_[i]);
   }
 }
-#endif
 
 TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
   NetEqOutputType type;
@@ -1172,7 +1183,11 @@ TEST_F(NetEqBgnTestFade, RunTest) {
 }
 
 #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
-TEST_F(NetEqDecodingTest, SyncPacketInsert) {
+#define MAYBE_SyncPacketInsert SyncPacketInsert
+#else
+#define MAYBE_SyncPacketInsert DISABLED_SyncPacketInsert
+#endif
+TEST_F(NetEqDecodingTest, MAYBE_SyncPacketInsert) {
   WebRtcRTPHeader rtp_info;
   uint32_t receive_timestamp = 0;
   // For the readability use the following payloads instead of the defaults of
@@ -1251,7 +1266,6 @@ TEST_F(NetEqDecodingTest, SyncPacketInsert) {
   --rtp_info.header.ssrc;
   EXPECT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
 }
-#endif
 
 // First insert several noise like packets, then sync-packets. Decoding all
 // packets should not produce error, statistics should not show any packet loss
diff --git a/webrtc/modules/modules_unittests.isolate b/webrtc/modules/modules_unittests.isolate
index d988821af0..c81197c77e 100644
--- a/webrtc/modules/modules_unittests.isolate
+++ b/webrtc/modules/modules_unittests.isolate
@@ -11,6 +11,9 @@
       'variables': {
         'files': [
           '<(DEPTH)/data/audio_processing/output_data_fixed.pb',
+          '<(DEPTH)/resources/audio_coding/neteq4_network_stats_android.dat',
+          '<(DEPTH)/resources/audio_coding/neteq4_rtcp_stats_android.dat',
+          '<(DEPTH)/resources/audio_coding/neteq4_universal_ref_android.pcm',
         ],
       },
     }],
diff --git a/webrtc/test/testsupport/fileutils.cc b/webrtc/test/testsupport/fileutils.cc
index 15abf5c517..4ff0034dea 100644
--- a/webrtc/test/testsupport/fileutils.cc
+++ b/webrtc/test/testsupport/fileutils.cc
@@ -222,6 +222,9 @@ std::string ResourcePath(std::string name, std::string extension) {
 #ifdef WEBRTC_MAC
   platform = "mac";
 #endif  // WEBRTC_MAC
+#ifdef WEBRTC_ANDROID
+  platform = "android";
+#endif  // WEBRTC_ANDROID
 
 #ifdef WEBRTC_ARCH_64_BITS
   std::string architecture = "64";