Enable clang::find_bad_constructs for audio_coding (part 1/2).

This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.

Bug: webrtc:9251, webrtc:163
Change-Id: I6a7d4964723a5e195189aac30a83d9e924e61dd7
Reviewed-on: https://webrtc-review.googlesource.com/89743
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24053}

modules/audio_coding/BUILD.gn

@@ -1119,11 +1119,6 @@ rtc_source_set("neteq_tools_minimal") {
     "neteq/tools/packet_source.h",
   ]
 
-  if (!build_with_chromium && is_clang) {
-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
-  }
-
   deps = [
     ":neteq",
     "../..:typedefs",
@@ -1510,31 +1505,22 @@ if (rtc_include_tests) {
       "neteq/audio_decoder_unittest.cc",
     ]
 
-    if (!build_with_chromium && is_clang) {
-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
-    }
-
-    deps = [
-      "../../test:fileutils",
-    ]
-
     defines = neteq_defines
 
-    deps += audio_coding_deps
-    deps += [
-      ":ilbc",
-      ":isac",
-      ":isac_fix",
-      ":neteq",
-      ":neteq_tools",
-      "../../api/audio_codecs:audio_codecs_api",
-      "../../api/audio_codecs/opus:audio_encoder_opus",
-      "../../common_audio",
-      "../../rtc_base:protobuf_utils",
-      "../../test:test_main",
-      "//testing/gtest",
-    ]
+    deps = [
+             ":ilbc",
+             ":isac",
+             ":isac_fix",
+             ":neteq",
+             ":neteq_tools",
+             "../../test:fileutils",
+             "../../api/audio_codecs:audio_codecs_api",
+             "../../api/audio_codecs/opus:audio_encoder_opus",
+             "../../common_audio",
+             "../../rtc_base:protobuf_utils",
+             "../../test:test_main",
+             "//testing/gtest",
+           ] + audio_coding_deps
 
     data = audio_decoder_unittests_resources
 
@@ -1545,7 +1531,7 @@ if (rtc_include_tests) {
     if (is_ios) {
       deps += [ ":audio_decoder_unittests_bundle_data" ]
     }
-  }  # audio_decoder_unittests
+  }
 
   if (rtc_enable_protobuf) {
     proto_library("neteq_unittest_proto") {
@@ -1777,11 +1763,6 @@ if (rtc_include_tests) {
       "../../system_wrappers:system_wrappers_default",
       "//testing/gtest",
     ]
-
-    if (!build_with_chromium && is_clang) {
-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
-    }
   }
 
   rtc_executable("neteq_opus_quality_test") {
@@ -2020,11 +2001,6 @@ if (rtc_include_tests) {
       "../../test:test_main",
       "//testing/gtest",
     ]
-
-    if (!build_with_chromium && is_clang) {
-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
-    }
   }
 
   rtc_source_set("audio_coding_unittests") {

modules/audio_coding/codecs/opus/opus_fec_test.cc

@@ -38,8 +38,8 @@ class OpusFecTest : public TestWithParam<coding_param> {
  protected:
   OpusFecTest();
 
-  virtual void SetUp();
-  virtual void TearDown();
+  void SetUp() override;
+  void TearDown() override;
 
   virtual void EncodeABlock();
 

modules/audio_coding/neteq/audio_decoder_unittest.cc

@@ -100,16 +100,16 @@ class AudioDecoderTest : public ::testing::Test {
         payload_type_(17),
         decoder_(NULL) {}
 
-  virtual ~AudioDecoderTest() {}
+  ~AudioDecoderTest() override {}
 
-  virtual void SetUp() {
+  void SetUp() override {
     if (audio_encoder_)
       codec_input_rate_hz_ = audio_encoder_->SampleRateHz();
     // Create arrays.
     ASSERT_GT(data_length_, 0u) << "The test must set data_length_ > 0";
   }
 
-  virtual void TearDown() {
+  void TearDown() override {
     delete decoder_;
     decoder_ = NULL;
   }

modules/audio_coding/neteq/tools/encode_neteq_input.cc

@@ -27,6 +27,8 @@ EncodeNetEqInput::EncodeNetEqInput(std::unique_ptr<Generator> generator,
   CreatePacket();
 }
 
+EncodeNetEqInput::~EncodeNetEqInput() = default;
+
 absl::optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
   RTC_DCHECK(packet_data_);
   return static_cast<int64_t>(packet_data_->time_ms);
@@ -50,6 +52,10 @@ void EncodeNetEqInput::AdvanceOutputEvent() {
   next_output_event_ms_ += kOutputPeriodMs;
 }
 
+bool EncodeNetEqInput::ended() const {
+  return next_output_event_ms_ <= input_duration_ms_;
+}
+
 absl::optional<RTPHeader> EncodeNetEqInput::NextHeader() const {
   RTC_DCHECK(packet_data_);
   return packet_data_->header;

modules/audio_coding/neteq/tools/encode_neteq_input.h

@@ -35,6 +35,7 @@ class EncodeNetEqInput : public NetEqInput {
   EncodeNetEqInput(std::unique_ptr<Generator> generator,
                    std::unique_ptr<AudioEncoder> encoder,
                    int64_t input_duration_ms);
+  ~EncodeNetEqInput() override;
 
   absl::optional<int64_t> NextPacketTime() const override;
 
@@ -44,9 +45,7 @@ class EncodeNetEqInput : public NetEqInput {
 
   void AdvanceOutputEvent() override;
 
-  bool ended() const override {
-    return next_output_event_ms_ <= input_duration_ms_;
-  }
+  bool ended() const override;
 
   absl::optional<RTPHeader> NextHeader() const override;
 

modules/audio_coding/neteq/tools/neteq_input.cc

@@ -15,6 +15,9 @@
 namespace webrtc {
 namespace test {
 
+NetEqInput::PacketData::PacketData() = default;
+NetEqInput::PacketData::~PacketData() = default;
+
 std::string NetEqInput::PacketData::ToString() const {
   std::stringstream ss;
   ss << "{"
@@ -34,6 +37,8 @@ TimeLimitedNetEqInput::TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input,
       start_time_ms_(input_->NextEventTime()),
       duration_ms_(duration_ms) {}
 
+TimeLimitedNetEqInput::~TimeLimitedNetEqInput() = default;
+
 absl::optional<int64_t> TimeLimitedNetEqInput::NextPacketTime() const {
   return ended_ ? absl::nullopt : input_->NextPacketTime();
 }

modules/audio_coding/neteq/tools/neteq_input.h

@@ -28,6 +28,8 @@ namespace test {
 class NetEqInput {
  public:
   struct PacketData {
+    PacketData();
+    ~PacketData();
     std::string ToString() const;
 
     RTPHeader header;
@@ -84,6 +86,7 @@ class NetEqInput {
 class TimeLimitedNetEqInput : public NetEqInput {
  public:
   TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input, int64_t duration_ms);
+  ~TimeLimitedNetEqInput() override;
   absl::optional<int64_t> NextPacketTime() const override;
   absl::optional<int64_t> NextOutputEventTime() const override;
   std::unique_ptr<PacketData> PopPacket() override;

modules/audio_coding/neteq/tools/neteq_test.cc

@@ -46,6 +46,8 @@ NetEqTest::NetEqTest(const NetEq::Config& config,
   RegisterExternalDecoders(ext_codecs);
 }
 
+NetEqTest::~NetEqTest() = default;
+
 int64_t NetEqTest::Run() {
   const int64_t start_time_ms = *input_->NextEventTime();
   int64_t time_now_ms = start_time_ms;

modules/audio_coding/neteq/tools/neteq_test.h

@@ -82,7 +82,7 @@ class NetEqTest {
             std::unique_ptr<AudioSink> output,
             Callbacks callbacks);
 
-  ~NetEqTest() = default;
+  ~NetEqTest();
 
   // Runs the test. Returns the duration of the produced audio in ms.
   int64_t Run();

modules/audio_coding/neteq/tools/rtp_file_source.h

@@ -39,7 +39,7 @@ class RtpFileSource : public PacketSource {
   static bool ValidRtpDump(const std::string& file_name);
   static bool ValidPcap(const std::string& file_name);
 
-  virtual ~RtpFileSource();
+  ~RtpFileSource() override;
 
   // Registers an RTP header extension and binds it to |id|.
   virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);