ClangTidy fixes for call/

These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.

Bug: webrtc:10410
Change-Id: I08ff36bd689fa7c3716c8e7017bd571cc9f09f35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127642
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27125}

call/bitrate_allocator_unittest.cc

@@ -19,7 +19,6 @@
 
 using ::testing::_;
 using ::testing::NiceMock;
-using ::testing::StrictMock;
 
 namespace webrtc {
 // Emulating old interface for test suite compatibility.

call/bitrate_estimator_tests.cc

@@ -66,7 +66,7 @@ class LogObserver {
         num_popped++;
         EXPECT_TRUE(a.find(b) != std::string::npos) << a << " != " << b;
       }
-      if (expected_log_lines_.size() <= 0) {
+      if (expected_log_lines_.empty()) {
         if (num_popped > 0) {
           done_.Set();
         }

call/call.cc

@@ -1092,16 +1092,16 @@ void Call::UpdateAggregateNetworkState() {
   bool have_video = false;
   {
     ReadLockScoped read_lock(*send_crit_);
-    if (audio_send_ssrcs_.size() > 0)
+    if (!audio_send_ssrcs_.empty())
       have_audio = true;
-    if (video_send_ssrcs_.size() > 0)
+    if (!video_send_ssrcs_.empty())
       have_video = true;
   }
   {
     ReadLockScoped read_lock(*receive_crit_);
-    if (audio_receive_streams_.size() > 0)
+    if (!audio_receive_streams_.empty())
       have_audio = true;
-    if (video_receive_streams_.size() > 0)
+    if (!video_receive_streams_.empty())
       have_video = true;
   }
 

call/call_factory.cc

@@ -25,7 +25,7 @@ namespace webrtc {
 namespace {
 bool ParseConfigParam(std::string exp_name, int* field) {
   std::string group = field_trial::FindFullName(exp_name);
-  if (group == "")
+  if (group.empty())
     return false;
 
   return (sscanf(group.c_str(), "%d", field) == 1);

call/call_perf_tests.cc

@@ -637,7 +637,7 @@ void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
     // TODO(holmer): Run this with a timer instead of once per packet.
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       VideoSendStream::Stats stats = send_stream_->GetStats();
-      if (stats.substreams.size() > 0) {
+      if (!stats.substreams.empty()) {
         RTC_DCHECK_EQ(1, stats.substreams.size());
         int bitrate_kbps =
             stats.substreams.begin()->second.total_bitrate_bps / 1000;

call/fake_network_pipe_unittest.cc

@@ -20,9 +20,6 @@
 #include "test/gtest.h"
 
 using ::testing::_;
-using ::testing::AnyNumber;
-using ::testing::Invoke;
-using ::testing::Return;
 
 namespace webrtc {
 

call/rtp_transport_controller_send_interface.h

@@ -35,7 +35,6 @@ class TaskQueue;
 }  // namespace rtc
 namespace webrtc {
 
-class CallStats;
 class CallStatsObserver;
 class FrameEncryptorInterface;
 class TargetTransferRateObserver;

call/rtp_video_sender_unittest.cc

@@ -27,10 +27,7 @@
 #include "video/send_statistics_proxy.h"
 
 using ::testing::_;
-using ::testing::AnyNumber;
-using ::testing::Invoke;
 using ::testing::NiceMock;
-using ::testing::Return;
 using ::testing::SaveArg;
 using ::testing::Unused;