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GN: Add video_engine_tests

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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}
This commit is contained in:
Peter Boström
2016-06-14 12:52:54 +02:00
parent 075af92730
commit 0208322ee3
18 changed files with 122 additions and 17 deletions
Download Patch File Download Diff File Expand all files Collapse all files

18
webrtc/BUILD.gn
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@ -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" ]
}
}
}

20
webrtc/audio/BUILD.gn
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@ -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" ]
}
}
}

2
webrtc/audio/audio_receive_stream_unittest.cc
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@ -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);

23
webrtc/call/BUILD.gn
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@ -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" ]
}
}
}

3
webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
View File

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

4
webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
View File

@ -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;
}

5
webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
View File

@ -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,
int32_t(const uint64_t pictureID));
MOCK_METHOD1(SendRTCPReferencePictureSelection, int32_t(uint64_t pictureID));
MOCK_METHOD1(SendRTCPSliceLossIndication,
int32_t(const uint8_t pictureID));
MOCK_CONST_METHOD2(DataCountersRTP,

2
webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
View File

@ -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);

2
webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
View File

@ -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);

2
webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
View File

@ -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_; }

2
webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
View File

@ -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) {

2
webrtc/modules/video_coding/test/receiver_tests.h
View File

@ -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);
}

2
webrtc/modules/video_coding/test/vcm_payload_sink_factory.cc
View File

@ -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);
}

11
webrtc/test/BUILD.gn
View File

@ -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",

35
webrtc/video/BUILD.gn
View File

@ -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" ]
}
}
}

2
webrtc/video/rtp_stream_receiver.cc
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@ -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;

2
webrtc/video/rtp_stream_receiver.h
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@ -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;

2
webrtc/video/video_send_stream_tests.cc
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@ -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;