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Fix Chromium clang plugin warnings

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NOTRY=true
BUG=webrtc:163

Review-Url: https://codereview.webrtc.org/2286283002
Cr-Commit-Position: refs/heads/master@{#13965}
This commit is contained in:
kwiberg
2016-08-29 15:10:34 -07:00
committed by Commit bot
parent 83ffe453ec
commit 3cc8774935
6 changed files with 23 additions and 25 deletions
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2
webrtc/modules/audio_processing/test/aec_dump_based_simulator.h
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@ -29,7 +29,7 @@ class AecDumpBasedSimulator final : public AudioProcessingSimulator {
public:
explicit AecDumpBasedSimulator(const SimulationSettings& settings)
: AudioProcessingSimulator(settings) {}
virtual ~AecDumpBasedSimulator() {}
~AecDumpBasedSimulator() override {}
// Processes the messages in the aecdump file.
void Process() override;

10
webrtc/modules/audio_processing/test/audio_processing_simulator.cc
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@ -44,6 +44,10 @@ std::string GetIndexedOutputWavFilename(const std::string& wav_name,
} // namespace
SimulationSettings::SimulationSettings() = default;
SimulationSettings::SimulationSettings(const SimulationSettings&) = default;
SimulationSettings::~SimulationSettings() = default;
void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) {
RTC_CHECK_EQ(src.num_channels(), dest->num_channels_);
RTC_CHECK_EQ(src.num_frames(), dest->samples_per_channel_);
@ -55,6 +59,12 @@ void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) {
}
}
AudioProcessingSimulator::AudioProcessingSimulator(
const SimulationSettings& settings)
: settings_(settings) {}
AudioProcessingSimulator::~AudioProcessingSimulator() = default;
AudioProcessingSimulator::ScopedTimer::~ScopedTimer() {
int64_t interval = rtc::TimeNanos() - start_time_;
proc_time_->sum += interval;

8
webrtc/modules/audio_processing/test/audio_processing_simulator.h
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@ -28,6 +28,9 @@ namespace test {
// Holds all the parameters available for controlling the simulation.
struct SimulationSettings {
SimulationSettings();
SimulationSettings(const SimulationSettings&);
~SimulationSettings();
rtc::Optional<int> stream_delay;
rtc::Optional<int> stream_drift_samples;
rtc::Optional<int> output_sample_rate_hz;
@ -92,9 +95,8 @@ class AudioProcessingSimulator {
public:
static const int kChunksPerSecond = 1000 / AudioProcessing::kChunkSizeMs;
explicit AudioProcessingSimulator(const SimulationSettings& settings)
: settings_(settings) {}
virtual ~AudioProcessingSimulator() {}
explicit AudioProcessingSimulator(const SimulationSettings& settings);
virtual ~AudioProcessingSimulator();
// Processes the data in the input.
virtual void Process() = 0;

5
webrtc/modules/audio_processing/test/wav_based_simulator.cc
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@ -16,6 +16,11 @@
namespace webrtc {
namespace test {
WavBasedSimulator::WavBasedSimulator(const SimulationSettings& settings)
: AudioProcessingSimulator(settings) {}
WavBasedSimulator::~WavBasedSimulator() = default;
std::vector<WavBasedSimulator::SimulationEventType>
WavBasedSimulator::GetDefaultEventChain() const {
std::vector<WavBasedSimulator::SimulationEventType> call_chain(2);

5
webrtc/modules/audio_processing/test/wav_based_simulator.h
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@ -23,9 +23,8 @@ namespace test {
// Used to perform an audio processing simulation from wav files.
class WavBasedSimulator final : public AudioProcessingSimulator {
public:
explicit WavBasedSimulator(const SimulationSettings& settings)
: AudioProcessingSimulator(settings) {}
virtual ~WavBasedSimulator() {}
explicit WavBasedSimulator(const SimulationSettings& settings);
~WavBasedSimulator() override;
// Processes the WAV input.
void Process() override;