Replace use of ASSERT in test code.
In top level test functions, replaced with gtest ASSERT_*. In helper
methods in main test files, replaced with EXPECT_* or RTC_DCHECK on a
case-by-case basis.
In separate mock/fake classes used by tests (which might be of some
use also in tests of third-party applications), ASSERT was replaced
with RTC_CHECK, using
git grep -l ' ASSERT(' | grep -v common.h | \
xargs sed -i 's/ ASSERT(/ RTC_CHECK(/'
followed by additional includes of base/checks.h in affected files,
and git cl format.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2622413005
Cr-Commit-Position: refs/heads/master@{#16150}
diff --git a/webrtc/api/rtcstatscollector_unittest.cc b/webrtc/api/rtcstatscollector_unittest.cc
index f8efb24..0ba9abf 100644
--- a/webrtc/api/rtcstatscollector_unittest.cc
+++ b/webrtc/api/rtcstatscollector_unittest.cc
@@ -790,22 +790,22 @@
expected_outbound_video_codec.codec = "video/VP8";
expected_outbound_video_codec.clock_rate = 1340;
- ASSERT(report->Get(expected_inbound_audio_codec.id()));
+ ASSERT_TRUE(report->Get(expected_inbound_audio_codec.id()));
EXPECT_EQ(expected_inbound_audio_codec,
report->Get(expected_inbound_audio_codec.id())->cast_to<
RTCCodecStats>());
- ASSERT(report->Get(expected_outbound_audio_codec.id()));
+ ASSERT_TRUE(report->Get(expected_outbound_audio_codec.id()));
EXPECT_EQ(expected_outbound_audio_codec,
report->Get(expected_outbound_audio_codec.id())->cast_to<
RTCCodecStats>());
- ASSERT(report->Get(expected_inbound_video_codec.id()));
+ ASSERT_TRUE(report->Get(expected_inbound_video_codec.id()));
EXPECT_EQ(expected_inbound_video_codec,
report->Get(expected_inbound_video_codec.id())->cast_to<
RTCCodecStats>());
- ASSERT(report->Get(expected_outbound_video_codec.id()));
+ ASSERT_TRUE(report->Get(expected_outbound_video_codec.id()));
EXPECT_EQ(expected_outbound_video_codec,
report->Get(expected_outbound_video_codec.id())->cast_to<
RTCCodecStats>());
@@ -1618,7 +1618,7 @@
expected_audio.jitter = 4.5;
expected_audio.fraction_lost = 5.5;
- ASSERT(report->Get(expected_audio.id()));
+ ASSERT_TRUE(report->Get(expected_audio.id()));
const RTCInboundRTPStreamStats& audio = report->Get(
expected_audio.id())->cast_to<RTCInboundRTPStreamStats>();
EXPECT_EQ(audio, expected_audio);
@@ -1703,7 +1703,7 @@
expected_video.fraction_lost = 4.5;
expected_video.frames_decoded = 8;
- ASSERT(report->Get(expected_video.id()));
+ ASSERT_TRUE(report->Get(expected_video.id()));
const RTCInboundRTPStreamStats& video = report->Get(
expected_video.id())->cast_to<RTCInboundRTPStreamStats>();
EXPECT_EQ(video, expected_video);
@@ -1776,7 +1776,7 @@
expected_audio.bytes_sent = 3;
expected_audio.round_trip_time = 4.5;
- ASSERT(report->Get(expected_audio.id()));
+ ASSERT_TRUE(report->Get(expected_audio.id()));
const RTCOutboundRTPStreamStats& audio = report->Get(
expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>();
EXPECT_EQ(audio, expected_audio);
@@ -1859,7 +1859,7 @@
expected_video.frames_encoded = 8;
expected_video.qp_sum = 16;
- ASSERT(report->Get(expected_video.id()));
+ ASSERT_TRUE(report->Get(expected_video.id()));
const RTCOutboundRTPStreamStats& video = report->Get(
expected_video.id())->cast_to<RTCOutboundRTPStreamStats>();
EXPECT_EQ(video, expected_video);
@@ -1943,7 +1943,7 @@
expected_audio.bytes_sent = 3;
// |expected_audio.round_trip_time| should be undefined.
- ASSERT(report->Get(expected_audio.id()));
+ ASSERT_TRUE(report->Get(expected_audio.id()));
const RTCOutboundRTPStreamStats& audio = report->Get(
expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>();
EXPECT_EQ(audio, expected_audio);
@@ -1965,7 +1965,7 @@
// |expected_video.round_trip_time| should be undefined.
// |expected_video.qp_sum| should be undefined.
- ASSERT(report->Get(expected_video.id()));
+ ASSERT_TRUE(report->Get(expected_video.id()));
const RTCOutboundRTPStreamStats& video = report->Get(
expected_video.id())->cast_to<RTCOutboundRTPStreamStats>();
EXPECT_EQ(video, expected_video);
diff --git a/webrtc/api/statscollector_unittest.cc b/webrtc/api/statscollector_unittest.cc
index 30bb2b8..f72e355 100644
--- a/webrtc/api/statscollector_unittest.cc
+++ b/webrtc/api/statscollector_unittest.cc
@@ -618,7 +618,7 @@
StatsReports* reports) {
// A track can't have both sender report and recv report at the same time
// for now, this might change in the future though.
- ASSERT((voice_sender_info == NULL) ^ (voice_receiver_info == NULL));
+ EXPECT_TRUE((voice_sender_info == NULL) ^ (voice_receiver_info == NULL));
// Instruct the session to return stats containing the transport channel.
InitSessionStats(vc_name);
@@ -1315,7 +1315,7 @@
uint32_t priority = 1000;
cricket::Candidate c;
- ASSERT(c.id().length() > 0);
+ EXPECT_GT(c.id().length(), 0u);
c.set_type(cricket::LOCAL_PORT_TYPE);
c.set_protocol(cricket::UDP_PROTOCOL_NAME);
c.set_address(local_address);
@@ -1325,7 +1325,7 @@
EXPECT_EQ("Cand-" + c.id(), report_id);
c = cricket::Candidate();
- ASSERT(c.id().length() > 0);
+ EXPECT_GT(c.id().length(), 0u);
c.set_type(cricket::PRFLX_PORT_TYPE);
c.set_protocol(cricket::UDP_PROTOCOL_NAME);
c.set_address(remote_address);
diff --git a/webrtc/api/test/fakeaudiocapturemodule.cc b/webrtc/api/test/fakeaudiocapturemodule.cc
index f118967..c0b761f 100644
--- a/webrtc/api/test/fakeaudiocapturemodule.cc
+++ b/webrtc/api/test/fakeaudiocapturemodule.cc
@@ -639,7 +639,7 @@
}
void FakeAudioCaptureModule::StartProcessP() {
- ASSERT(process_thread_->IsCurrent());
+ RTC_CHECK(process_thread_->IsCurrent());
if (started_) {
// Already started.
return;
@@ -648,7 +648,7 @@
}
void FakeAudioCaptureModule::ProcessFrameP() {
- ASSERT(process_thread_->IsCurrent());
+ RTC_CHECK(process_thread_->IsCurrent());
if (!started_) {
next_frame_time_ = rtc::TimeMillis();
started_ = true;
@@ -673,7 +673,7 @@
}
void FakeAudioCaptureModule::ReceiveFrameP() {
- ASSERT(process_thread_->IsCurrent());
+ RTC_CHECK(process_thread_->IsCurrent());
{
rtc::CritScope cs(&crit_callback_);
if (!audio_callback_) {
@@ -689,7 +689,7 @@
&elapsed_time_ms, &ntp_time_ms) != 0) {
RTC_NOTREACHED();
}
- ASSERT(nSamplesOut == kNumberSamples);
+ RTC_CHECK(nSamplesOut == kNumberSamples);
}
// The SetBuffer() function ensures that after decoding, the audio buffer
// should contain samples of similar magnitude (there is likely to be some
@@ -704,7 +704,7 @@
}
void FakeAudioCaptureModule::SendFrameP() {
- ASSERT(process_thread_->IsCurrent());
+ RTC_CHECK(process_thread_->IsCurrent());
rtc::CritScope cs(&crit_callback_);
if (!audio_callback_) {
return;
diff --git a/webrtc/api/test/fakedatachannelprovider.h b/webrtc/api/test/fakedatachannelprovider.h
index 3404ac1..3e796a3 100644
--- a/webrtc/api/test/fakedatachannelprovider.h
+++ b/webrtc/api/test/fakedatachannelprovider.h
@@ -12,6 +12,7 @@
#define WEBRTC_API_TEST_FAKEDATACHANNELPROVIDER_H_
#include "webrtc/api/datachannel.h"
+#include "webrtc/base/checks.h"
class FakeDataChannelProvider : public webrtc::DataChannelProviderInterface {
public:
@@ -25,7 +26,7 @@
bool SendData(const cricket::SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
cricket::SendDataResult* result) override {
- ASSERT(ready_to_send_ && transport_available_);
+ RTC_CHECK(ready_to_send_ && transport_available_);
if (send_blocked_) {
*result = cricket::SDR_BLOCK;
return false;
@@ -41,7 +42,8 @@
}
bool ConnectDataChannel(webrtc::DataChannel* data_channel) override {
- ASSERT(connected_channels_.find(data_channel) == connected_channels_.end());
+ RTC_CHECK(connected_channels_.find(data_channel) ==
+ connected_channels_.end());
if (!transport_available_) {
return false;
}
@@ -51,13 +53,14 @@
}
void DisconnectDataChannel(webrtc::DataChannel* data_channel) override {
- ASSERT(connected_channels_.find(data_channel) != connected_channels_.end());
+ RTC_CHECK(connected_channels_.find(data_channel) !=
+ connected_channels_.end());
LOG(LS_INFO) << "DataChannel disconnected " << data_channel;
connected_channels_.erase(data_channel);
}
void AddSctpDataStream(int sid) override {
- ASSERT(sid >= 0);
+ RTC_CHECK(sid >= 0);
if (!transport_available_) {
return;
}
@@ -66,7 +69,7 @@
}
void RemoveSctpDataStream(int sid) override {
- ASSERT(sid >= 0);
+ RTC_CHECK(sid >= 0);
send_ssrcs_.erase(sid);
recv_ssrcs_.erase(sid);
}
@@ -99,7 +102,7 @@
// Set true to emulate the transport ReadyToSendData signal when the transport
// becomes writable for the first time.
void set_ready_to_send(bool ready) {
- ASSERT(transport_available_);
+ RTC_CHECK(transport_available_);
ready_to_send_ = ready;
if (ready) {
std::set<webrtc::DataChannel*>::iterator it;
diff --git a/webrtc/api/test/mockpeerconnectionobservers.h b/webrtc/api/test/mockpeerconnectionobservers.h
index 23647f6..1f000af 100644
--- a/webrtc/api/test/mockpeerconnectionobservers.h
+++ b/webrtc/api/test/mockpeerconnectionobservers.h
@@ -17,6 +17,7 @@
#include <string>
#include "webrtc/api/datachannelinterface.h"
+#include "webrtc/base/checks.h"
namespace webrtc {
@@ -109,7 +110,7 @@
virtual ~MockStatsObserver() {}
virtual void OnComplete(const StatsReports& reports) {
- ASSERT(!called_);
+ RTC_CHECK(!called_);
called_ = true;
stats_.Clear();
stats_.number_of_reports = reports.size();
@@ -143,37 +144,37 @@
double timestamp() const { return stats_.timestamp; }
int AudioOutputLevel() const {
- ASSERT(called_);
+ RTC_CHECK(called_);
return stats_.audio_output_level;
}
int AudioInputLevel() const {
- ASSERT(called_);
+ RTC_CHECK(called_);
return stats_.audio_input_level;
}
int BytesReceived() const {
- ASSERT(called_);
+ RTC_CHECK(called_);
return stats_.bytes_received;
}
int BytesSent() const {
- ASSERT(called_);
+ RTC_CHECK(called_);
return stats_.bytes_sent;
}
int AvailableReceiveBandwidth() const {
- ASSERT(called_);
+ RTC_CHECK(called_);
return stats_.available_receive_bandwidth;
}
std::string DtlsCipher() const {
- ASSERT(called_);
+ RTC_CHECK(called_);
return stats_.dtls_cipher;
}
std::string SrtpCipher() const {
- ASSERT(called_);
+ RTC_CHECK(called_);
return stats_.srtp_cipher;
}
diff --git a/webrtc/api/webrtcsession_unittest.cc b/webrtc/api/webrtcsession_unittest.cc
index 3e1a583..ca09315 100644
--- a/webrtc/api/webrtcsession_unittest.cc
+++ b/webrtc/api/webrtcsession_unittest.cc
@@ -1450,10 +1450,10 @@
bool VerifyNoCNCodecs(const cricket::ContentInfo* content) {
const cricket::ContentDescription* description = content->description;
- ASSERT(description != NULL);
+ RTC_CHECK(description != NULL);
const cricket::AudioContentDescription* audio_content_desc =
static_cast<const cricket::AudioContentDescription*>(description);
- ASSERT(audio_content_desc != NULL);
+ RTC_CHECK(audio_content_desc != NULL);
for (size_t i = 0; i < audio_content_desc->codecs().size(); ++i) {
if (audio_content_desc->codecs()[i].name == "CN")
return false;
@@ -1463,7 +1463,7 @@
void CreateDataChannel() {
webrtc::InternalDataChannelInit dci;
- ASSERT(session_.get());
+ RTC_CHECK(session_.get());
dci.reliable = session_->data_channel_type() == cricket::DCT_SCTP;
data_channel_ = DataChannel::Create(
session_.get(), session_->data_channel_type(), "datachannel", dci);
@@ -3082,7 +3082,7 @@
session_->video_rtp_transport_channel());
cricket::BaseChannel* voice_channel = session_->voice_channel();
- ASSERT(voice_channel != NULL);
+ ASSERT_TRUE(voice_channel != NULL);
// Checks if one of the transport channels contains a connection using a given
// port.
diff --git a/webrtc/base/network_unittest.cc b/webrtc/base/network_unittest.cc
index ab5df73..2c05b05 100644
--- a/webrtc/base/network_unittest.cc
+++ b/webrtc/base/network_unittest.cc
@@ -10,6 +10,7 @@
#include "webrtc/base/network.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/nethelpers.h"
#include "webrtc/base/networkmonitor.h"
#include <memory>
@@ -103,7 +104,7 @@
AdapterType GetAdapterType(BasicNetworkManager& network_manager) {
BasicNetworkManager::NetworkList list;
network_manager.GetNetworks(&list);
- ASSERT(list.size() == 1u);
+ RTC_CHECK(list.size() == 1u);
return list[0]->type();
}
diff --git a/webrtc/base/optionsfile_unittest.cc b/webrtc/base/optionsfile_unittest.cc
index bc3d38e..4eaf801 100644
--- a/webrtc/base/optionsfile_unittest.cc
+++ b/webrtc/base/optionsfile_unittest.cc
@@ -10,6 +10,7 @@
#include <memory>
+#include "webrtc/base/checks.h"
#include "webrtc/base/fileutils.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/optionsfile.h"
@@ -46,7 +47,7 @@
public:
MAYBE_OptionsFileTest() {
Pathname dir;
- ASSERT(Filesystem::GetTemporaryFolder(dir, true, NULL));
+ RTC_CHECK(Filesystem::GetTemporaryFolder(dir, true, NULL));
test_file_ = Filesystem::TempFilename(dir, ".testfile");
OpenStore();
}
diff --git a/webrtc/base/sharedexclusivelock_unittest.cc b/webrtc/base/sharedexclusivelock_unittest.cc
index 3886fb2..701405a 100644
--- a/webrtc/base/sharedexclusivelock_unittest.cc
+++ b/webrtc/base/sharedexclusivelock_unittest.cc
@@ -10,6 +10,7 @@
#include <memory>
+#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/event.h"
@@ -60,9 +61,9 @@
private:
virtual void OnMessage(Message* message) {
- ASSERT(rtc::Thread::Current() == worker_thread_.get());
- ASSERT(message != NULL);
- ASSERT(message->message_id == kMsgRead);
+ RTC_CHECK(rtc::Thread::Current() == worker_thread_.get());
+ RTC_CHECK(message != NULL);
+ RTC_CHECK(message->message_id == kMsgRead);
TypedMessageData<int*>* message_data =
static_cast<TypedMessageData<int*>*>(message->pdata);
@@ -90,9 +91,9 @@
private:
virtual void OnMessage(Message* message) {
- ASSERT(rtc::Thread::Current() == worker_thread_.get());
- ASSERT(message != NULL);
- ASSERT(message->message_id == kMsgWrite);
+ RTC_CHECK(rtc::Thread::Current() == worker_thread_.get());
+ RTC_CHECK(message != NULL);
+ RTC_CHECK(message->message_id == kMsgWrite);
TypedMessageData<int>* message_data =
static_cast<TypedMessageData<int>*>(message->pdata);
diff --git a/webrtc/base/sslstreamadapter_unittest.cc b/webrtc/base/sslstreamadapter_unittest.cc
index 9e156c0..fa4ed6d 100644
--- a/webrtc/base/sslstreamadapter_unittest.cc
+++ b/webrtc/base/sslstreamadapter_unittest.cc
@@ -15,6 +15,7 @@
#include <string>
#include "webrtc/base/bufferqueue.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/ssladapter.h"
@@ -383,7 +384,7 @@
// Make sure we simulate a reliable network for TLS.
// This is just a check to make sure that people don't write wrong
// tests.
- ASSERT((mtu_ == 1460) && (loss_ == 0) && (lose_first_packet_ == 0));
+ RTC_CHECK((mtu_ == 1460) && (loss_ == 0) && (lose_first_packet_ == 0));
}
if (!identities_set_)
@@ -420,7 +421,7 @@
// Make sure we simulate a reliable network for TLS.
// This is just a check to make sure that people don't write wrong
// tests.
- ASSERT((mtu_ == 1460) && (loss_ == 0) && (lose_first_packet_ == 0));
+ RTC_CHECK((mtu_ == 1460) && (loss_ == 0) && (lose_first_packet_ == 0));
}
// Start the handshake
diff --git a/webrtc/base/testutils.h b/webrtc/base/testutils.h
index 34b7606..332857d 100644
--- a/webrtc/base/testutils.h
+++ b/webrtc/base/testutils.h
@@ -28,6 +28,7 @@
#include <vector>
#include "webrtc/base/arraysize.h"
#include "webrtc/base/asyncsocket.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/nethelpers.h"
@@ -177,7 +178,7 @@
va_start(args, format);
char buffer[1024];
size_t len = vsprintfn(buffer, sizeof(buffer), format, args);
- ASSERT(len < sizeof(buffer) - 1);
+ RTC_CHECK(len < sizeof(buffer) - 1);
va_end(args);
QueueData(buffer, len);
}
@@ -297,7 +298,7 @@
va_start(args, format);
char buffer[1024];
size_t len = vsprintfn(buffer, sizeof(buffer), format, args);
- ASSERT(len < sizeof(buffer) - 1);
+ RTC_CHECK(len < sizeof(buffer) - 1);
va_end(args);
QueueData(buffer, len);
}
diff --git a/webrtc/p2p/base/dtlstransportchannel_unittest.cc b/webrtc/p2p/base/dtlstransportchannel_unittest.cc
index bff2e7d..4eba26d 100644
--- a/webrtc/p2p/base/dtlstransportchannel_unittest.cc
+++ b/webrtc/p2p/base/dtlstransportchannel_unittest.cc
@@ -14,6 +14,7 @@
#include "webrtc/p2p/base/dtlstransportchannel.h"
#include "webrtc/p2p/base/faketransportcontroller.h"
#include "webrtc/p2p/base/packettransportinterface.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/common.h"
#include "webrtc/base/dscp.h"
#include "webrtc/base/gunit.h"
@@ -78,7 +79,7 @@
return certificate_;
}
void SetupSrtp() {
- ASSERT(certificate_);
+ EXPECT_TRUE(certificate_ != nullptr);
use_dtls_srtp_ = true;
}
void SetupMaxProtocolVersion(rtc::SSLProtocolVersion version) {
@@ -300,7 +301,7 @@
}
void SendPackets(size_t channel, size_t size, size_t count, bool srtp) {
- ASSERT(channel < channels_.size());
+ RTC_CHECK(channel < channels_.size());
std::unique_ptr<char[]> packet(new char[size]);
size_t sent = 0;
do {
@@ -324,7 +325,7 @@
}
int SendInvalidSrtpPacket(size_t channel, size_t size) {
- ASSERT(channel < channels_.size());
+ RTC_CHECK(channel < channels_.size());
std::unique_ptr<char[]> packet(new char[size]);
// Fill the packet with 0 to form an invalid SRTP packet.
memset(packet.get(), 0, size);
diff --git a/webrtc/p2p/base/p2ptransportchannel_unittest.cc b/webrtc/p2p/base/p2ptransportchannel_unittest.cc
index 1f262db..e5fa8a1 100644
--- a/webrtc/p2p/base/p2ptransportchannel_unittest.cc
+++ b/webrtc/p2p/base/p2ptransportchannel_unittest.cc
@@ -19,6 +19,7 @@
#include "webrtc/p2p/base/teststunserver.h"
#include "webrtc/p2p/base/testturnserver.h"
#include "webrtc/p2p/client/basicportallocator.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/dscp.h"
#include "webrtc/base/fakeclock.h"
#include "webrtc/base/fakenetwork.h"
@@ -2056,7 +2057,7 @@
class P2PTransportChannelSameNatTest : public P2PTransportChannelTestBase {
protected:
void ConfigureEndpoints(Config nat_type, Config config1, Config config2) {
- ASSERT(nat_type >= NAT_FULL_CONE && nat_type <= NAT_SYMMETRIC);
+ RTC_CHECK(nat_type >= NAT_FULL_CONE && nat_type <= NAT_SYMMETRIC);
rtc::NATSocketServer::Translator* outer_nat =
nat()->AddTranslator(kPublicAddrs[0], kNatAddrs[0],
static_cast<rtc::NATType>(nat_type - NAT_FULL_CONE));
@@ -2066,7 +2067,7 @@
}
void ConfigureEndpoint(rtc::NATSocketServer::Translator* nat,
int endpoint, Config config) {
- ASSERT(config <= NAT_SYMMETRIC);
+ RTC_CHECK(config <= NAT_SYMMETRIC);
if (config == OPEN) {
AddAddress(endpoint, kPrivateAddrs[endpoint]);
nat->AddClient(kPrivateAddrs[endpoint]);
diff --git a/webrtc/p2p/base/port_unittest.cc b/webrtc/p2p/base/port_unittest.cc
index 88ed7f4..38012a6 100644
--- a/webrtc/p2p/base/port_unittest.cc
+++ b/webrtc/p2p/base/port_unittest.cc
@@ -1300,7 +1300,7 @@
ch1.CreateConnection(GetCandidate(port2));
int64_t after_created = rtc::TimeMillis();
Connection* conn = ch1.conn();
- ASSERT(conn != nullptr);
+ ASSERT_NE(conn, nullptr);
// It is not dead if it is after MIN_CONNECTION_LIFETIME but not pruned.
conn->UpdateState(after_created + MIN_CONNECTION_LIFETIME + 1);
rtc::Thread::Current()->ProcessMessages(0);
@@ -1318,7 +1318,7 @@
// Create a connection again and receive a ping.
ch1.CreateConnection(GetCandidate(port2));
conn = ch1.conn();
- ASSERT(conn != nullptr);
+ ASSERT_NE(conn, nullptr);
int64_t before_last_receiving = rtc::TimeMillis();
conn->ReceivedPing();
int64_t after_last_receiving = rtc::TimeMillis();
diff --git a/webrtc/p2p/base/turnport_unittest.cc b/webrtc/p2p/base/turnport_unittest.cc
index edad812..69b25ea 100644
--- a/webrtc/p2p/base/turnport_unittest.cc
+++ b/webrtc/p2p/base/turnport_unittest.cc
@@ -22,6 +22,7 @@
#include "webrtc/p2p/base/udpport.h"
#include "webrtc/base/asynctcpsocket.h"
#include "webrtc/base/buffer.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/dscp.h"
#include "webrtc/base/fakeclock.h"
#include "webrtc/base/firewallsocketserver.h"
@@ -163,7 +164,7 @@
}
virtual void OnMessage(rtc::Message* msg) {
- ASSERT(msg->message_id == MSG_TESTFINISH);
+ RTC_CHECK(msg->message_id == MSG_TESTFINISH);
if (msg->message_id == MSG_TESTFINISH)
test_finish_ = true;
}
@@ -273,7 +274,7 @@
void CreateSharedTurnPort(const std::string& username,
const std::string& password,
const ProtocolAddress& server_address) {
- ASSERT(server_address.proto == PROTO_UDP);
+ RTC_CHECK(server_address.proto == PROTO_UDP);
if (!socket_) {
socket_.reset(socket_factory_.CreateUdpSocket(
diff --git a/webrtc/pc/channel_unittest.cc b/webrtc/pc/channel_unittest.cc
index 110ea04..dd5eaa1 100644
--- a/webrtc/pc/channel_unittest.cc
+++ b/webrtc/pc/channel_unittest.cc
@@ -12,6 +12,7 @@
#include "webrtc/base/array_view.h"
#include "webrtc/base/buffer.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/fakeclock.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
@@ -1268,8 +1269,8 @@
// Test that we properly send SRTP with RTCP in both directions.
// You can pass in DTLS and/or RTCP_MUX as flags.
void SendSrtpToSrtp(int flags1_in = 0, int flags2_in = 0) {
- ASSERT((flags1_in & ~(RTCP_MUX | DTLS | GCM_CIPHER)) == 0);
- ASSERT((flags2_in & ~(RTCP_MUX | DTLS | GCM_CIPHER)) == 0);
+ RTC_CHECK((flags1_in & ~(RTCP_MUX | DTLS | GCM_CIPHER)) == 0);
+ RTC_CHECK((flags2_in & ~(RTCP_MUX | DTLS | GCM_CIPHER)) == 0);
int flags1 = SECURE | flags1_in;
int flags2 = SECURE | flags2_in;
diff --git a/webrtc/pc/mediasession_unittest.cc b/webrtc/pc/mediasession_unittest.cc
index 45de0d2..cd2ea23 100644
--- a/webrtc/pc/mediasession_unittest.cc
+++ b/webrtc/pc/mediasession_unittest.cc
@@ -12,6 +12,7 @@
#include <string>
#include <vector>
+#include "webrtc/base/checks.h"
#include "webrtc/base/fakesslidentity.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/messagedigest.h"
@@ -451,10 +452,10 @@
bool VerifyNoCNCodecs(const cricket::ContentInfo* content) {
const cricket::ContentDescription* description = content->description;
- ASSERT(description != NULL);
+ RTC_CHECK(description != NULL);
const cricket::AudioContentDescription* audio_content_desc =
static_cast<const cricket::AudioContentDescription*>(description);
- ASSERT(audio_content_desc != NULL);
+ RTC_CHECK(audio_content_desc != NULL);
for (size_t i = 0; i < audio_content_desc->codecs().size(); ++i) {
if (audio_content_desc->codecs()[i].name == "CN")
return false;
