Fix Chromium clang plugin warnings
NOTRY=true
BUG=webrtc:163
Review-Url: https://codereview.webrtc.org/2288153002
Cr-Commit-Position: refs/heads/master@{#13964}
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index 349a797..6730fd5 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -239,11 +239,6 @@
# TODO(jschuh): Bug 1348: fix this warning.
configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
- if (is_clang) {
- # Suppress warnings from the Chromium Clang plugins (bugs.webrtc.org/163).
- configs -= [ "//build/config/clang:find_bad_constructs" ]
- }
-
deps += [
"../../base:rtc_base_approved",
"../../common_audio",
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation.cc b/webrtc/modules/audio_processing/aec/echo_cancellation.cc
index 716da38..c26b1a0 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation.cc
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation.cc
@@ -28,6 +28,9 @@
namespace webrtc {
+Aec::Aec() = default;
+Aec::~Aec() = default;
+
// Measured delays [ms]
// Device Chrome GTP
// MacBook Air 10
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation.h b/webrtc/modules/audio_processing/aec/echo_cancellation.h
index 8e5e52c..1047113 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation.h
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation.h
@@ -65,6 +65,9 @@
class ApmDataDumper;
typedef struct Aec {
+ Aec();
+ ~Aec();
+
std::unique_ptr<ApmDataDumper> data_dumper;
int delayCtr;
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 011325f..ff475c9 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -278,6 +278,20 @@
return MaybeInitialize(processing_config);
}
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+
+AudioProcessingImpl::ApmDebugDumpThreadState::ApmDebugDumpThreadState()
+ : event_msg(new audioproc::Event()) {}
+
+AudioProcessingImpl::ApmDebugDumpThreadState::~ApmDebugDumpThreadState() {}
+
+AudioProcessingImpl::ApmDebugDumpState::ApmDebugDumpState()
+ : debug_file(FileWrapper::Create()) {}
+
+AudioProcessingImpl::ApmDebugDumpState::~ApmDebugDumpState() {}
+
+#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
+
// Calls InitializeLocked() if any of the audio parameters have changed from
// their current values (needs to be called while holding the crit_render_lock).
int AudioProcessingImpl::MaybeInitialize(
@@ -1524,4 +1538,28 @@
}
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
+AudioProcessingImpl::ApmCaptureState::ApmCaptureState(
+ bool transient_suppressor_enabled,
+ const std::vector<Point>& array_geometry,
+ SphericalPointf target_direction)
+ : aec_system_delay_jumps(-1),
+ delay_offset_ms(0),
+ was_stream_delay_set(false),
+ last_stream_delay_ms(0),
+ last_aec_system_delay_ms(0),
+ stream_delay_jumps(-1),
+ output_will_be_muted(false),
+ key_pressed(false),
+ transient_suppressor_enabled(transient_suppressor_enabled),
+ array_geometry(array_geometry),
+ target_direction(target_direction),
+ fwd_proc_format(kSampleRate16kHz),
+ split_rate(kSampleRate16kHz) {}
+
+AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
+
+AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
+
+AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
+
} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index 4b9011d..87e2224 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -45,7 +45,7 @@
explicit AudioProcessingImpl(const Config& config);
// AudioProcessingImpl takes ownership of beamformer.
AudioProcessingImpl(const Config& config, NonlinearBeamformer* beamformer);
- virtual ~AudioProcessingImpl();
+ ~AudioProcessingImpl() override;
int Initialize() override;
int Initialize(int input_sample_rate_hz,
int output_sample_rate_hz,
@@ -133,7 +133,8 @@
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// State for the debug dump.
struct ApmDebugDumpThreadState {
- ApmDebugDumpThreadState() : event_msg(new audioproc::Event()) {}
+ ApmDebugDumpThreadState();
+ ~ApmDebugDumpThreadState();
std::unique_ptr<audioproc::Event> event_msg; // Protobuf message.
std::string event_str; // Memory for protobuf serialization.
@@ -142,7 +143,8 @@
};
struct ApmDebugDumpState {
- ApmDebugDumpState() : debug_file(FileWrapper::Create()) {}
+ ApmDebugDumpState();
+ ~ApmDebugDumpState();
// Number of bytes that can still be written to the log before the maximum
// size is reached. A value of <= 0 indicates that no limit is used.
int64_t num_bytes_left_for_log_ = -1;
@@ -287,20 +289,8 @@
struct ApmCaptureState {
ApmCaptureState(bool transient_suppressor_enabled,
const std::vector<Point>& array_geometry,
- SphericalPointf target_direction)
- : aec_system_delay_jumps(-1),
- delay_offset_ms(0),
- was_stream_delay_set(false),
- last_stream_delay_ms(0),
- last_aec_system_delay_ms(0),
- stream_delay_jumps(-1),
- output_will_be_muted(false),
- key_pressed(false),
- transient_suppressor_enabled(transient_suppressor_enabled),
- array_geometry(array_geometry),
- target_direction(target_direction),
- fwd_proc_format(kSampleRate16kHz),
- split_rate(kSampleRate16kHz) {}
+ SphericalPointf target_direction);
+ ~ApmCaptureState();
int aec_system_delay_jumps;
int delay_offset_ms;
bool was_stream_delay_set;
@@ -342,6 +332,8 @@
} capture_nonlocked_;
struct ApmRenderState {
+ ApmRenderState();
+ ~ApmRenderState();
std::unique_ptr<AudioConverter> render_converter;
std::unique_ptr<AudioBuffer> render_audio;
} render_ GUARDED_BY(crit_render_);
diff --git a/webrtc/modules/audio_processing/echo_cancellation_impl.h b/webrtc/modules/audio_processing/echo_cancellation_impl.h
index f67ef7b..0f0945f 100644
--- a/webrtc/modules/audio_processing/echo_cancellation_impl.h
+++ b/webrtc/modules/audio_processing/echo_cancellation_impl.h
@@ -28,7 +28,7 @@
public:
EchoCancellationImpl(rtc::CriticalSection* crit_render,
rtc::CriticalSection* crit_capture);
- virtual ~EchoCancellationImpl();
+ ~EchoCancellationImpl() override;
int ProcessRenderAudio(const AudioBuffer* audio);
int ProcessCaptureAudio(AudioBuffer* audio, int stream_delay_ms);
diff --git a/webrtc/modules/audio_processing/echo_control_mobile_impl.h b/webrtc/modules/audio_processing/echo_control_mobile_impl.h
index 97e2407..791a4d5 100644
--- a/webrtc/modules/audio_processing/echo_control_mobile_impl.h
+++ b/webrtc/modules/audio_processing/echo_control_mobile_impl.h
@@ -29,7 +29,7 @@
EchoControlMobileImpl(rtc::CriticalSection* crit_render,
rtc::CriticalSection* crit_capture);
- virtual ~EchoControlMobileImpl();
+ ~EchoControlMobileImpl() override;
int ProcessRenderAudio(const AudioBuffer* audio);
int ProcessCaptureAudio(AudioBuffer* audio, int stream_delay_ms);
diff --git a/webrtc/modules/audio_processing/level_controller/signal_classifier.cc b/webrtc/modules/audio_processing/level_controller/signal_classifier.cc
index 4f2d998..c9a07e7 100644
--- a/webrtc/modules/audio_processing/level_controller/signal_classifier.cc
+++ b/webrtc/modules/audio_processing/level_controller/signal_classifier.cc
@@ -93,6 +93,12 @@
} // namespace
+SignalClassifier::FrameExtender::FrameExtender(size_t frame_size,
+ size_t extended_frame_size)
+ : x_old_(extended_frame_size - frame_size, 0.f) {}
+
+SignalClassifier::FrameExtender::~FrameExtender() = default;
+
void SignalClassifier::FrameExtender::ExtendFrame(
rtc::ArrayView<const float> x,
rtc::ArrayView<float> x_extended) {
diff --git a/webrtc/modules/audio_processing/level_controller/signal_classifier.h b/webrtc/modules/audio_processing/level_controller/signal_classifier.h
index cfa0fc5..2afa724 100644
--- a/webrtc/modules/audio_processing/level_controller/signal_classifier.h
+++ b/webrtc/modules/audio_processing/level_controller/signal_classifier.h
@@ -37,8 +37,8 @@
private:
class FrameExtender {
public:
- FrameExtender(size_t frame_size, size_t extended_frame_size)
- : x_old_(extended_frame_size - frame_size, 0.f) {}
+ FrameExtender(size_t frame_size, size_t extended_frame_size);
+ ~FrameExtender();
void ExtendFrame(rtc::ArrayView<const float> x,
rtc::ArrayView<float> x_extended);
