GN: Add video_engine_tests

Adds separate source_sets for the video_engine_tests subtargets inside
audio, call and video and merges them together into video_engine_tests.

BUG=webrtc:5949
R=kjellander@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/2064523002 .

Cr-Commit-Position: refs/heads/master@{#13127}

webrtc/BUILD.gn

@@ -593,4 +593,22 @@ if (rtc_include_tests) {
       ldflags = [ "-ObjC" ]
     }
   }
+
+  # TODO(pbos): Rename test suite, this is no longer "just" for video targets.
+  test("video_engine_tests") {
+    testonly = true
+    deps = [
+      "audio:audio_tests",
+      "call:call_tests",
+      "modules/video_capture",
+      "test:test_common",
+      "test:test_main",
+      "video:video_tests",
+    ]
+    if (is_clang) {
+      # Suppress warnings from the Chromium Clang plugin.
+      # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+      configs -= [ "//build/config/clang:find_bad_constructs" ]
+    }
+  }
 }

webrtc/audio/BUILD.gn

@@ -35,3 +35,23 @@ source_set("audio") {
     "../voice_engine",
   ]
 }
+if (rtc_include_tests) {
+  source_set("audio_tests") {
+    testonly = true
+    sources = [
+      "audio_receive_stream_unittest.cc",
+      "audio_send_stream_unittest.cc",
+      "audio_state_unittest.cc",
+    ]
+    deps = [
+      ":audio",
+      "//testing/gmock",
+      "//testing/gtest",
+    ]
+    if (is_clang) {
+      # Suppress warnings from the Chromium Clang plugin.
+      # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+      configs -= [ "//build/config/clang:find_bad_constructs" ]
+    }
+  }
+}

webrtc/audio/audio_receive_stream_unittest.cc

@@ -187,7 +187,7 @@ void BuildOneByteExtension(std::vector<uint8_t>::iterator it,
   const size_t kExtensionDataLength = kOneByteExtensionLength - 1;
   uint32_t shifted_value = extension_value
                            << (8 * (kExtensionDataLength - value_length));
-  *it = (id << 4) + (value_length - 1);
+  *it = (id << 4) + (static_cast<uint8_t>(value_length) - 1);
   ++it;
   ByteWriter<uint32_t, kExtensionDataLength>::WriteBigEndian(&(*it),
                                                              shifted_value);

webrtc/call/BUILD.gn

@@ -35,3 +35,26 @@ source_set("call") {
     "../video",
   ]
 }
+
+if (rtc_include_tests) {
+  source_set("call_tests") {
+    testonly = true
+    sources = [
+      "bitrate_allocator_unittest.cc",
+      "bitrate_estimator_tests.cc",
+      "call_unittest.cc",
+      "packet_injection_tests.cc",
+      "ringbuffer_unittest.cc",
+    ]
+    deps = [
+      ":call",
+      "//testing/gmock",
+      "//testing/gtest",
+    ]
+    if (is_clang) {
+      # Suppress warnings from the Chromium Clang plugin.
+      # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+      configs -= [ "//build/config/clang:find_bad_constructs" ]
+    }
+  }
+}

webrtc/modules/rtp_rtcp/include/rtp_rtcp.h

@@ -416,8 +416,7 @@ class RtpRtcp : public Module {
     /*
     *    Good state of RTP receiver inform sender
     */
-    virtual int32_t SendRTCPReferencePictureSelection(
+    virtual int32_t SendRTCPReferencePictureSelection(uint64_t pictureID) = 0;
-        const uint64_t pictureID) = 0;
 
     /*
     *    Send a RTCP Slice Loss Indication (SLI)

webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h

@@ -189,7 +189,7 @@ class RtpData {
   virtual ~RtpData() {}
 
   virtual int32_t OnReceivedPayloadData(const uint8_t* payloadData,
-                                        const size_t payloadSize,
+                                        size_t payloadSize,
                                         const WebRtcRTPHeader* rtpHeader) = 0;
 
   virtual bool OnRecoveredPacket(const uint8_t* packet,
@@ -351,7 +351,7 @@ class NullRtpData : public RtpData {
   virtual ~NullRtpData() {}
 
   int32_t OnReceivedPayloadData(const uint8_t* payloadData,
-                                const size_t payloadSize,
+                                size_t payloadSize,
                                 const WebRtcRTPHeader* rtpHeader) override {
     return 0;
   }

webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h

@@ -28,7 +28,7 @@ class MockRtpData : public RtpData {
  public:
   MOCK_METHOD3(OnReceivedPayloadData,
                int32_t(const uint8_t* payloadData,
-                       const size_t payloadSize,
+                       size_t payloadSize,
                        const WebRtcRTPHeader* rtpHeader));
 
   MOCK_METHOD2(OnRecoveredPacket,
@@ -168,8 +168,7 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_METHOD1(SendRTCP, int32_t(RTCPPacketType packetType));
   MOCK_METHOD1(SendCompoundRTCP,
                int32_t(const std::set<RTCPPacketType>& packetTypes));
-  MOCK_METHOD1(SendRTCPReferencePictureSelection,
+  MOCK_METHOD1(SendRTCPReferencePictureSelection, int32_t(uint64_t pictureID));
-      int32_t(const uint64_t pictureID));
   MOCK_METHOD1(SendRTCPSliceLossIndication,
       int32_t(const uint8_t pictureID));
   MOCK_CONST_METHOD2(DataCountersRTP,

webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc

@@ -41,7 +41,7 @@ class VerifyingRtxReceiver : public NullRtpData {
 
   int32_t OnReceivedPayloadData(
       const uint8_t* data,
-      const size_t size,
+      size_t size,
       const webrtc::WebRtcRTPHeader* rtp_header) override {
     if (!sequence_numbers_.empty())
       EXPECT_EQ(kTestSsrc, rtp_header->header.ssrc);

webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc

@@ -69,7 +69,7 @@ bool LoopBackTransport::SendRtcp(const uint8_t* data, size_t len) {
 
 int32_t TestRtpReceiver::OnReceivedPayloadData(
     const uint8_t* payload_data,
-    const size_t payload_size,
+    size_t payload_size,
     const webrtc::WebRtcRTPHeader* rtp_header) {
   EXPECT_LE(payload_size, sizeof(payload_data_));
   memcpy(payload_data_, payload_data, payload_size);

webrtc/modules/rtp_rtcp/test/testAPI/test_api.h

@@ -55,7 +55,7 @@ class TestRtpReceiver : public NullRtpData {
  public:
   int32_t OnReceivedPayloadData(
       const uint8_t* payload_data,
-      const size_t payload_size,
+      size_t payload_size,
       const webrtc::WebRtcRTPHeader* rtp_header) override;
 
   const uint8_t* payload_data() const { return payload_data_; }

webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

@@ -28,7 +28,7 @@ class VerifyingAudioReceiver : public NullRtpData {
  public:
   int32_t OnReceivedPayloadData(
       const uint8_t* payloadData,
-      const size_t payloadSize,
+      size_t payloadSize,
       const webrtc::WebRtcRTPHeader* rtpHeader) override {
     if (rtpHeader->header.payloadType == 98 ||
         rtpHeader->header.payloadType == 99) {

webrtc/modules/video_coding/test/receiver_tests.h

@@ -29,7 +29,7 @@ class RtpDataCallback : public webrtc::NullRtpData {
 
   int32_t OnReceivedPayloadData(
       const uint8_t* payload_data,
-      const size_t payload_size,
+      size_t payload_size,
       const webrtc::WebRtcRTPHeader* rtp_header) override {
     return vcm_->IncomingPacket(payload_data, payload_size, *rtp_header);
   }

webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc

@@ -47,7 +47,7 @@ class VcmPayloadSinkFactory::VcmPayloadSink : public PayloadSinkInterface,
 
   // PayloadSinkInterface
   int32_t OnReceivedPayloadData(const uint8_t* payload_data,
-                                const size_t payload_size,
+                                size_t payload_size,
                                 const WebRtcRTPHeader* rtp_header) override {
     return vcm_->IncomingPacket(payload_data, payload_size, *rtp_header);
   }

webrtc/test/BUILD.gn

@@ -187,6 +187,8 @@ source_set("test_support") {
     "testsupport/frame_writer.cc",
     "testsupport/frame_writer.h",
     "testsupport/iosfileutils.mm",
+    "testsupport/metrics/video_metrics.cc",
+    "testsupport/metrics/video_metrics.h",
     "testsupport/mock/mock_frame_reader.h",
     "testsupport/mock/mock_frame_writer.h",
     "testsupport/packet_reader.cc",
@@ -199,11 +201,18 @@ source_set("test_support") {
 
   deps = [
     "../base:gtest_prod",
+    "../common_video",
     "../system_wrappers",
     "//testing/gmock",
     "//testing/gtest",
   ]
 
+  if (is_clang) {
+    # Suppress warnings from the Chromium Clang plugin.
+    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+    configs -= [ "//build/config/clang:find_bad_constructs" ]
+  }
+
   if (is_ios) {
     configs += [ "//build/config/compiler:enable_arc" ]
   }
@@ -272,6 +281,7 @@ test("test_support_unittests") {
     "channel_transport/udp_socket_manager_unittest.cc",
     "channel_transport/udp_socket_wrapper_unittest.cc",
     "channel_transport/udp_transport_unittest.cc",
+    "common_unittest.cc",
     "fake_network_pipe_unittest.cc",
     "frame_generator_unittest.cc",
     "rtp_file_reader_unittest.cc",
@@ -280,6 +290,7 @@ test("test_support_unittests") {
     "testsupport/fileutils_unittest.cc",
     "testsupport/frame_reader_unittest.cc",
     "testsupport/frame_writer_unittest.cc",
+    "testsupport/metrics/video_metrics_unittest.cc",
     "testsupport/packet_reader_unittest.cc",
     "testsupport/perf_test_unittest.cc",
     "testsupport/unittest_utils.h",

webrtc/video/BUILD.gn

@@ -7,6 +7,7 @@
 # be found in the AUTHORS file in the root of the source tree.
 
 import("../build/webrtc.gni")
+import("//testing/test.gni")
 
 source_set("video") {
   sources = [
@@ -79,3 +80,37 @@ source_set("video") {
     "../voice_engine",
   ]
 }
+
+if (rtc_include_tests) {
+  # TODO(pbos): Rename test suite.
+  source_set("video_tests") {
+    testonly = true
+    sources = [
+      "call_stats_unittest.cc",
+      "encoder_state_feedback_unittest.cc",
+      "end_to_end_tests.cc",
+      "overuse_frame_detector_unittest.cc",
+      "payload_router_unittest.cc",
+      "report_block_stats_unittest.cc",
+      "send_delay_stats_unittest.cc",
+      "send_statistics_proxy_unittest.cc",
+      "stats_counter_unittest.cc",
+      "stream_synchronization_unittest.cc",
+      "video_capture_input_unittest.cc",
+      "video_decoder_unittest.cc",
+      "video_encoder_unittest.cc",
+      "video_send_stream_tests.cc",
+      "vie_remb_unittest.cc",
+    ]
+    deps = [
+      ":video",
+      "//testing/gmock",
+      "//testing/gtest",
+    ]
+    if (is_clang) {
+      # Suppress warnings from the Chromium Clang plugin.
+      # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+      configs -= [ "//build/config/clang:find_bad_constructs" ]
+    }
+  }
+}

webrtc/video/rtp_stream_receiver.cc

@@ -226,7 +226,7 @@ RtpReceiver* RtpStreamReceiver::GetRtpReceiver() const {
 
 int32_t RtpStreamReceiver::OnReceivedPayloadData(
     const uint8_t* payload_data,
-    const size_t payload_size,
+    size_t payload_size,
     const WebRtcRTPHeader* rtp_header) {
   RTC_DCHECK(video_receiver_);
   WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;

webrtc/video/rtp_stream_receiver.h

@@ -85,7 +85,7 @@ class RtpStreamReceiver : public RtpData, public RtpFeedback,
 
   // Implements RtpData.
   int32_t OnReceivedPayloadData(const uint8_t* payload_data,
-                                const size_t payload_size,
+                                size_t payload_size,
                                 const WebRtcRTPHeader* rtp_header) override;
   bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
 

webrtc/video/video_send_stream_tests.cc

@@ -2118,7 +2118,7 @@ class Vp9HeaderObserver : public test::SendTest {
     EXPECT_TRUE(vp9.spatial_layer_resolution_present);  // Y:1
     size_t expected_width = encoder_config_.streams[0].width;
     size_t expected_height = encoder_config_.streams[0].height;
-    for (int i = vp9.num_spatial_layers - 1; i >= 0; --i) {
+    for (int i = static_cast<int>(vp9.num_spatial_layers) - 1; i >= 0; --i) {
       EXPECT_EQ(expected_width, vp9.width[i]);    // WIDTH
       EXPECT_EQ(expected_height, vp9.height[i]);  // HEIGHT
       expected_width /= 2;