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webrtc/src//59e1464fcd51ea5367f1c8dfdedea84f6271fe52^!/.
commit
59e1464fcd51ea5367f1c8dfdedea84f6271fe52
[log]
author
Danil Chapovalov <danilchap@webrtc.org>
Tue Sep 03 10:41:41 2019
committer
Commit Bot <commit-bot@chromium.org>
Wed Sep 04 07:38:20 2019
tree
d64ecb08d004eafed6cd67f22607b728b318f45e
parent
38350b1ef26330f3caebe84d4d935671d4e26720 [diff]
Fix 28 ClangTidy - Readability findings in modules/rtp_rtcp/

These fixes are automatically created by various analysis tools, but have been manually triggered to be applied.
 * the 'empty' method should be used to check for emptiness instead of 'size' (3 times)
 * using decl 'Return' is unused (4 times)
 * using decl '_' is unused (3 times)
 * using decl 'DoAll' is unused (2 times)
 * using decl 'SetArgPointee' is unused
 * using decl 'Dlrr' is unused
 * using decl 'IsEmpty' is unused
 * redundant get() call on smart pointer
 * using decl 'Invoke' is unused (2 times)
 * using decl 'SizeIs' is unused (3 times)
 * using decl 'make_tuple' is unused
 * using decl 'NiceMock' is unused
 * using decl 'SaveArg' is unused (2 times)
 * using decl 'AtLeast' is unused
 * using decl 'ElementsAre' is unused
 * using decl 'Gt' is unused

Bug: None
Change-Id: I97658fb0e94620b8319d7c3da29b15e27ec23188
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151133
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29056}

diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.cc b/modules/rtp_rtcp/include/rtp_rtcp_defines.cc
index 7f8efca..16f571f 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.cc
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.cc

@@ -33,12 +33,12 @@
 }  // namespace
 
 bool IsLegalMidName(absl::string_view name) {
-  return (name.size() <= kMidRsidMaxSize && name.size() > 0 &&
+  return (name.size() <= kMidRsidMaxSize && !name.empty() &&
           absl::c_all_of(name, IsTokenChar));
 }
 
 bool IsLegalRsidName(absl::string_view name) {
-  return (name.size() <= kMidRsidMaxSize && name.size() > 0 &&
+  return (name.size() <= kMidRsidMaxSize && !name.empty() &&
           absl::c_all_of(name, isalnum));
 }
 

diff --git a/modules/rtp_rtcp/source/flexfec_receiver_unittest.cc b/modules/rtp_rtcp/source/flexfec_receiver_unittest.cc
index 3d77d7b..be64bac 100644
--- a/modules/rtp_rtcp/source/flexfec_receiver_unittest.cc
+++ b/modules/rtp_rtcp/source/flexfec_receiver_unittest.cc

@@ -27,7 +27,6 @@
 using ::testing::_;
 using ::testing::Args;
 using ::testing::ElementsAreArray;
-using ::testing::Return;
 
 using test::fec::FlexfecPacketGenerator;
 using Packet = ForwardErrorCorrection::Packet;

diff --git a/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc b/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
index 9f42f57..c9b9434d 100644
--- a/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
+++ b/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc

@@ -14,11 +14,6 @@
 #include "test/gmock.h"
 #include "test/gtest.h"
 
-using ::testing::_;
-using ::testing::DoAll;
-using ::testing::Return;
-using ::testing::SetArgPointee;
-
 namespace webrtc {
 
 static const int64_t kTestRtt = 10;

diff --git a/modules/rtp_rtcp/source/rtcp_packet/extended_reports_unittest.cc b/modules/rtp_rtcp/source/rtcp_packet/extended_reports_unittest.cc
index a302a5b..7c50c01 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/extended_reports_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/extended_reports_unittest.cc

@@ -19,7 +19,6 @@
 using ::testing::ElementsAreArray;
 using ::testing::make_tuple;
 using ::testing::SizeIs;
-using webrtc::rtcp::Dlrr;
 using webrtc::rtcp::ExtendedReports;
 using webrtc::rtcp::ReceiveTimeInfo;
 using webrtc::rtcp::Rrtr;

diff --git a/modules/rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc b/modules/rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc
index a320f23..1bac808 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc

@@ -15,7 +15,6 @@
 #include "test/rtcp_packet_parser.h"
 
 using ::testing::ElementsAreArray;
-using ::testing::IsEmpty;
 using ::testing::make_tuple;
 using webrtc::rtcp::TmmbItem;
 using webrtc::rtcp::Tmmbr;

diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc
index c06a428..4c1a241 100644
--- a/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/modules/rtp_rtcp/source/rtcp_sender.cc

@@ -743,7 +743,7 @@
 
       BuilderFunc func = builder_it->second;
       std::unique_ptr<rtcp::RtcpPacket> packet = (this->*func)(context);
-      if (packet.get() == nullptr)
+      if (packet == nullptr)
         return -1;
       // If there is a BYE, don't append now - save it and append it
       // at the end later.

diff --git a/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc b/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
index fac1735..ebfb068 100644
--- a/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc

@@ -32,7 +32,6 @@
 
 using ::testing::_;
 using ::testing::ElementsAre;
-using ::testing::Invoke;
 using ::testing::Return;
 using ::testing::SizeIs;
 using ::testing::StrictMock;

diff --git a/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc b/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc
index 9ba0573..67b7cba 100644
--- a/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc

@@ -30,7 +30,6 @@
 using ::testing::Each;
 using ::testing::ElementsAreArray;
 using ::testing::Le;
-using ::testing::SizeIs;
 
 constexpr RtpPacketizer::PayloadSizeLimits kNoSizeLimits;
 

diff --git a/modules/rtp_rtcp/source/rtp_packet_unittest.cc b/modules/rtp_rtcp/source/rtp_packet_unittest.cc
index b583477..150dba7 100644
--- a/modules/rtp_rtcp/source/rtp_packet_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_packet_unittest.cc

@@ -23,7 +23,6 @@
 using ::testing::ElementsAre;
 using ::testing::ElementsAreArray;
 using ::testing::IsEmpty;
-using ::testing::make_tuple;
 
 constexpr int8_t kPayloadType = 100;
 constexpr uint32_t kSsrc = 0x12345678;

diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index d8ffd22..eaba788 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

@@ -694,7 +694,7 @@
   if (!rtp_sender_)
     return;
 
-  if (!rtp_sender_->StorePackets() || nack_sequence_numbers.size() == 0) {
+  if (!rtp_sender_->StorePackets() || nack_sequence_numbers.empty()) {
     return;
   }
   // Use RTT from RtcpRttStats class if provided.

diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index f55e4f8..3cec171 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc

@@ -29,11 +29,7 @@
 #include "test/rtcp_packet_parser.h"
 #include "test/rtp_header_parser.h"
 
-using ::testing::_;
 using ::testing::ElementsAre;
-using ::testing::NiceMock;
-using ::testing::Return;
-using ::testing::SaveArg;
 
 namespace webrtc {
 namespace {

diff --git a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
index 1dad5b7..1e5b38b 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc

@@ -33,7 +33,6 @@
 const uint8_t kAudioLevel = 0x5a;
 const uint64_t kStartTime = 123456789;
 
-using ::testing::_;
 using ::testing::ElementsAreArray;
 
 class LoopbackTransportTest : public webrtc::Transport {

diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index ad501df..7c4c374 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc

@@ -74,19 +74,12 @@
 
 using ::testing::_;
 using ::testing::AllOf;
-using ::testing::AtLeast;
-using ::testing::DoAll;
-using ::testing::ElementsAre;
 using ::testing::ElementsAreArray;
 using ::testing::Field;
-using ::testing::Gt;
-using ::testing::Invoke;
 using ::testing::NiceMock;
 using ::testing::Pointee;
 using ::testing::Property;
 using ::testing::Return;
-using ::testing::SaveArg;
-using ::testing::SizeIs;
 using ::testing::StrictMock;
 
 uint64_t ConvertMsToAbsSendTime(int64_t time_ms) {

diff --git a/modules/rtp_rtcp/source/source_tracker_unittest.cc b/modules/rtp_rtcp/source/source_tracker_unittest.cc
index 55ae4d1..99e0372 100644
--- a/modules/rtp_rtcp/source/source_tracker_unittest.cc
+++ b/modules/rtp_rtcp/source/source_tracker_unittest.cc

@@ -31,7 +31,6 @@
 using ::testing::ElementsAre;
 using ::testing::ElementsAreArray;
 using ::testing::IsEmpty;
-using ::testing::SizeIs;
 using ::testing::TestWithParam;
 using ::testing::Values;
 

diff --git a/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc b/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc
index 32f3bbb..8e41c1c 100644
--- a/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc
+++ b/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc

@@ -30,7 +30,6 @@
 using ::testing::_;
 using ::testing::Args;
 using ::testing::ElementsAreArray;
-using ::testing::Return;
 
 using test::fec::AugmentedPacket;
 using Packet = ForwardErrorCorrection::Packet;