diff --git a/webrtc/build/ios/tests/common_tests.json b/webrtc/build/ios/tests/common_tests.json
index 2d3f649f7d..767fc1233f 100644
--- a/webrtc/build/ios/tests/common_tests.json
+++ b/webrtc/build/ios/tests/common_tests.json
@@ -1,7 +1,31 @@
 {
   "tests": [
+    {
+      "app": "audio_decoder_unittests"
+    },
+    {
+      "app": "common_audio_unittests"
+    },
+    {
+      "app": "common_video_unittests"
+    },
+    {
+      "app": "modules_tests"
+    },
     {
       "app": "modules_unittests"
+    },
+    {
+      "app": "rtc_api_objc_tests"
+    },
+    {
+      "app": "rtc_pc_unittests"
+    },
+    {
+      "app": "system_wrappers_unittests"
+    },
+    {
+      "app": "voice_engine_unittests"
     }
   ]
 }
diff --git a/webrtc/common_video/common_video_unittests.gyp b/webrtc/common_video/common_video_unittests.gyp
index b5e892caf0..20203f1c9c 100644
--- a/webrtc/common_video/common_video_unittests.gyp
+++ b/webrtc/common_video/common_video_unittests.gyp
@@ -35,6 +35,11 @@
             '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
           ],
         }],
+        ['OS=="ios"', {
+          'mac_bundle_resources': [
+            '<(DEPTH)/resources/foreman_cif.yuv',
+          ],
+        }],
       ],
     },
   ],  # targets
diff --git a/webrtc/common_video/libyuv/libyuv_unittest.cc b/webrtc/common_video/libyuv/libyuv_unittest.cc
index e7cf0759f6..826fc0e16a 100644
--- a/webrtc/common_video/libyuv/libyuv_unittest.cc
+++ b/webrtc/common_video/libyuv/libyuv_unittest.cc
@@ -103,8 +103,8 @@ TestLibYuv::TestLibYuv()
 }
 
 void TestLibYuv::SetUp() {
-  const std::string input_file_name = webrtc::test::ProjectRootPath() +
-                                      "resources/foreman_cif.yuv";
+  const std::string input_file_name = webrtc::test::ResourcePath("foreman_cif",
+                                                                 "yuv");
   source_file_  = fopen(input_file_name.c_str(), "rb");
   ASSERT_TRUE(source_file_ != NULL) << "Cannot read file: "<<
                                        input_file_name << "\n";
diff --git a/webrtc/media/media_tests.gypi b/webrtc/media/media_tests.gypi
index 307033b467..fcac6b4549 100644
--- a/webrtc/media/media_tests.gypi
+++ b/webrtc/media/media_tests.gypi
@@ -164,6 +164,15 @@
           'sources!': [
             'sctp/sctpdataengine_unittest.cc',
           ],
+          'mac_bundle_resources': [
+            '<(DEPTH)/resources/media/captured-320x240-2s-48.frames',
+            '<(DEPTH)/resources/media/faces.1280x720_P420.yuv',
+            '<(DEPTH)/resources/media/faces_I420.jpg',
+            '<(DEPTH)/resources/media/faces_I422.jpg',
+            '<(DEPTH)/resources/media/faces_I444.jpg',
+            '<(DEPTH)/resources/media/faces_I411.jpg',
+            '<(DEPTH)/resources/media/faces_I400.jpg',
+          ],
         }],
       ],
     },  # target rtc_media_unittests
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 3a3b7bcb1a..58dd6c6680 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -96,9 +96,9 @@ double MseInputOutput(const std::vector<int16_t>& input,
 class AudioDecoderTest : public ::testing::Test {
  protected:
   AudioDecoderTest()
-      : input_audio_(webrtc::test::ProjectRootPath() +
-                         "resources/audio_coding/testfile32kHz.pcm",
-                     32000),
+      : input_audio_(
+            webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
+            32000),
         codec_input_rate_hz_(32000),  // Legacy default value.
         encoded_(NULL),
         frame_size_(0),
diff --git a/webrtc/modules/audio_coding/neteq/neteq.gypi b/webrtc/modules/audio_coding/neteq/neteq.gypi
index 50b8425c87..ead9586f5c 100644
--- a/webrtc/modules/audio_coding/neteq/neteq.gypi
+++ b/webrtc/modules/audio_coding/neteq/neteq.gypi
@@ -159,6 +159,11 @@
                 '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
               ],
             }],
+            ['OS=="ios"', {
+              'mac_bundle_resources': [
+                '<(DEPTH)/resources/audio_coding/testfile32kHz.pcm',
+              ],
+            }],
           ],
         }, # audio_decoder_unittests
 
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index b49a5d3dca..f1e88b7d41 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -107,6 +107,14 @@
                 '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
               ],
             }],
+            ['OS=="ios"', {
+              'mac_bundle_resources': [
+                '<(DEPTH)/resources/audio_coding/testfile32kHz.pcm',
+                '<(DEPTH)/resources/audio_coding/teststereo32kHz.pcm',
+                '<(DEPTH)/resources/foreman_cif.yuv',
+                '<(DEPTH)/resources/paris_qcif.yuv',
+              ],
+            }],
           ],
         },
       ],
diff --git a/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc b/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
index b164b7e04c..0a1ae6272e 100644
--- a/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
+++ b/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
@@ -81,7 +81,13 @@ void ReceivePackets(
   }
 }
 
-TEST(FecTest, FecTest) {
+// Too slow to finish before timeout on iOS. See webrtc:4755.
+#if defined(WEBRTC_IOS)
+#define MAYBE_FecTest DISABLED_FecTest
+#else
+#define MAYBE_FecTest FecTest
+#endif
+TEST(FecTest, MAYBE_FecTest) {
   // TODO(marpan): Split this function into subroutines/helper functions.
   enum { kMaxNumberMediaPackets = 48 };
   enum { kMaxNumberFecPackets = 48 };
diff --git a/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc b/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
index 79d75c903b..3df258a64d 100644
--- a/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
+++ b/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
@@ -84,7 +84,9 @@ struct RateControlMetrics {
 // Sequence used is foreman (CIF): may be better to use VGA for resize test.
 const int kCIFWidth = 352;
 const int kCIFHeight = 288;
+#if !defined(WEBRTC_IOS)
 const int kNbrFramesShort = 100;  // Some tests are run for shorter sequence.
+#endif
 const int kNbrFramesLong = 299;
 
 // Parameters from VP8 wrapper, which control target size of key frames.
@@ -621,6 +623,9 @@ TEST_F(VideoProcessorIntegrationTest, Process0PercentPacketLossH264) {
 
 #endif  // defined(WEBRTC_VIDEOPROCESSOR_H264_TESTS)
 
+// Fails on iOS. See webrtc:4755.
+#if !defined(WEBRTC_IOS)
+
 // VP9: Run with no packet loss and fixed bitrate. Quality should be very high.
 // One key frame (first frame only) in sequence. Setting |key_frame_interval|
 // to -1 below means no periodic key frames in test.
@@ -842,6 +847,8 @@ TEST_F(VideoProcessorIntegrationTest, Process10PercentPacketLoss) {
                          rc_metrics);
 }
 
+#endif  // !defined(WEBRTC_IOS)
+
 // The tests below are currently disabled for Android. For ARM, the encoder
 // uses |cpu_speed| = 12, as opposed to default |cpu_speed| <= 6 for x86,
 // which leads to significantly different quality. The quality and rate control
@@ -855,7 +862,8 @@ TEST_F(VideoProcessorIntegrationTest, Process10PercentPacketLoss) {
 // low to high to medium. Check that quality and encoder response to the new
 // target rate/per-frame bandwidth (for each rate update) is within limits.
 // One key frame (first frame only) in sequence.
-#if defined(WEBRTC_ANDROID)
+// Too slow to finish before timeout on iOS. See webrtc:4755.
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
 #define MAYBE_ProcessNoLossChangeBitRateVP8 \
   DISABLED_ProcessNoLossChangeBitRateVP8
 #else
@@ -892,7 +900,8 @@ TEST_F(VideoProcessorIntegrationTest, MAYBE_ProcessNoLossChangeBitRateVP8) {
 // for the rate control metrics can be lower. One key frame (first frame only).
 // Note: quality after update should be higher but we currently compute quality
 // metrics averaged over whole sequence run.
-#if defined(WEBRTC_ANDROID)
+// Too slow to finish before timeout on iOS. See webrtc:4755.
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
 #define MAYBE_ProcessNoLossChangeFrameRateFrameDropVP8 \
   DISABLED_ProcessNoLossChangeFrameRateFrameDropVP8
 #else
@@ -927,7 +936,8 @@ TEST_F(VideoProcessorIntegrationTest,
 
 // Run with no packet loss, at low bitrate. During this time we should've
 // resized once. Expect 2 key frames generated (first and one for resize).
-#if defined(WEBRTC_ANDROID)
+// Too slow to finish before timeout on iOS. See webrtc:4755.
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
 #define MAYBE_ProcessNoLossSpatialResizeFrameDropVP8 \
   DISABLED_ProcessNoLossSpatialResizeFrameDropVP8
 #else
@@ -961,7 +971,8 @@ TEST_F(VideoProcessorIntegrationTest,
 // encoding rate mismatch are applied to each layer.
 // No dropped frames in this test, and internal spatial resizer is off.
 // One key frame (first frame only) in sequence, so no spatial resizing.
-#if defined(WEBRTC_ANDROID)
+// Too slow to finish before timeout on iOS. See webrtc:4755.
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
 #define MAYBE_ProcessNoLossTemporalLayersVP8 \
   DISABLED_ProcessNoLossTemporalLayersVP8
 #else
diff --git a/webrtc/modules/video_coding/test/test_util.cc b/webrtc/modules/video_coding/test/test_util.cc
index 7ff663e395..8f5c55f9ba 100644
--- a/webrtc/modules/video_coding/test/test_util.cc
+++ b/webrtc/modules/video_coding/test/test_util.cc
@@ -26,7 +26,7 @@ CmdArgs::CmdArgs()
       width(352),
       height(288),
       rtt(0),
-      inputFile(webrtc::test::ProjectRootPath() + "/resources/foreman_cif.yuv"),
+      inputFile(webrtc::test::ResourcePath("foreman_cif", "yuv")),
       outputFile(webrtc::test::OutputPath() +
                  "video_coding_test_output_352x288.yuv") {}
 
diff --git a/webrtc/modules/video_coding/test/tester_main.cc b/webrtc/modules/video_coding/test/tester_main.cc
index 33ca82007d..1aa2780f0f 100644
--- a/webrtc/modules/video_coding/test/tester_main.cc
+++ b/webrtc/modules/video_coding/test/tester_main.cc
@@ -21,7 +21,7 @@ DEFINE_int32(width, 352, "Width in pixels of the frames in the input file.");
 DEFINE_int32(height, 288, "Height in pixels of the frames in the input file.");
 DEFINE_int32(rtt, 0, "RTT (round-trip time), in milliseconds.");
 DEFINE_string(input_filename,
-              webrtc::test::ProjectRootPath() + "/resources/foreman_cif.yuv",
+              webrtc::test::ResourcePath("foreman_cif", "yuv"),
               "Input file.");
 DEFINE_string(output_filename,
               webrtc::test::OutputPath() +
diff --git a/webrtc/test/testsupport/fileutils.cc b/webrtc/test/testsupport/fileutils.cc
index 4ff0034dea..2fab425a31 100644
--- a/webrtc/test/testsupport/fileutils.cc
+++ b/webrtc/test/testsupport/fileutils.cc
@@ -43,6 +43,7 @@ namespace test {
 
 #if defined(WEBRTC_IOS)
 // Defined in iosfileutils.mm.  No header file to discourage use elsewhere.
+std::string IOSOutputPath();
 std::string IOSResourcePath(std::string name, std::string extension);
 #endif
 
@@ -59,9 +60,12 @@ const char* kRootDirName = "/sdcard/";
 #else
 // The file we're looking for to identify the project root dir.
 const char* kProjectRootFileName = "DEPS";
+#if !defined(WEBRTC_IOS)
 const char* kOutputDirName = "out";
-const char* kFallbackPath = "./";
 #endif
+const char* kFallbackPath = "./";
+#endif  // !defined(WEBRTC_ANDROID)
+
 #if !defined(WEBRTC_IOS)
 const char* kResourcesDirName = "resources";
 #endif
@@ -140,6 +144,9 @@ std::string ProjectRootPath() {
 }
 
 std::string OutputPath() {
+#if defined(WEBRTC_IOS)
+  return IOSOutputPath();
+#else
   std::string path = ProjectRootPath();
   if (path == kCannotFindProjectRootDir) {
     return kFallbackPath;
@@ -149,6 +156,7 @@ std::string OutputPath() {
     return kFallbackPath;
   }
   return path + kPathDelimiter;
+#endif
 }
 
 std::string WorkingDir() {
diff --git a/webrtc/test/testsupport/iosfileutils.mm b/webrtc/test/testsupport/iosfileutils.mm
index f3615ed681..87b7397c79 100644
--- a/webrtc/test/testsupport/iosfileutils.mm
+++ b/webrtc/test/testsupport/iosfileutils.mm
@@ -54,6 +54,18 @@ std::string IOSResourcePath(std::string name, std::string extension) {
   }
 }
 
+// For iOS, we don't have access to the output directory. Return the path to the
+// temporary directory instead. This is mostly used by tests that need to write
+// output files to disk.
+std::string IOSOutputPath()  {
+  @autoreleasepool {
+    NSString* tempDir = NSTemporaryDirectory();
+    if (tempDir == nil)
+        tempDir = @"/tmp";
+    return StdStringFromNSString(tempDir);
+  }
+}
+
 }  // namespace test
 }  // namespace webrtc
 
diff --git a/webrtc/webrtc_tests.gypi b/webrtc/webrtc_tests.gypi
index 7b01ad1451..3563fee0f5 100644
--- a/webrtc/webrtc_tests.gypi
+++ b/webrtc/webrtc_tests.gypi
@@ -204,6 +204,12 @@
             '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
           ],
         }],
+        ['OS=="ios"', {
+          'mac_bundle_resources': [
+            '<(DEPTH)/resources/foreman_cif_short.yuv',
+            '<(DEPTH)/resources/voice_engine/audio_long16.pcm',
+          ],
+        }],
         ['enable_protobuf==1', {
           'defines': [
             'ENABLE_RTC_EVENT_LOG',