diff options
Diffstat (limited to 'chromium/third_party/webrtc/common_audio')
9 files changed, 30 insertions, 17 deletions
diff --git a/chromium/third_party/webrtc/common_audio/BUILD.gn b/chromium/third_party/webrtc/common_audio/BUILD.gn index 72eed1f0033..4077486d870 100644 --- a/chromium/third_party/webrtc/common_audio/BUILD.gn +++ b/chromium/third_party/webrtc/common_audio/BUILD.gn @@ -56,8 +56,8 @@ rtc_library("common_audio") { "../system_wrappers", "../system_wrappers:cpu_features_api", "third_party/ooura:fft_size_256", - "//third_party/abseil-cpp/absl/types:optional", ] + absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] defines = [] diff --git a/chromium/third_party/webrtc/common_audio/OWNERS b/chromium/third_party/webrtc/common_audio/OWNERS index 7c9c9af12a4..ba1c8b11f44 100644 --- a/chromium/third_party/webrtc/common_audio/OWNERS +++ b/chromium/third_party/webrtc/common_audio/OWNERS @@ -1,2 +1,3 @@ henrik.lundin@webrtc.org kwiberg@webrtc.org +peah@webrtc.org diff --git a/chromium/third_party/webrtc/common_audio/channel_buffer_unittest.cc b/chromium/third_party/webrtc/common_audio/channel_buffer_unittest.cc index 8ec42346d1c..a8b64891d6f 100644 --- a/chromium/third_party/webrtc/common_audio/channel_buffer_unittest.cc +++ b/chromium/third_party/webrtc/common_audio/channel_buffer_unittest.cc @@ -53,12 +53,12 @@ TEST(IFChannelBufferTest, SettingNumChannelsOfOneChannelBufferSetsTheOther) { } #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) -TEST(ChannelBufferTest, SetNumChannelsDeathTest) { +TEST(ChannelBufferDeathTest, SetNumChannelsDeathTest) { ChannelBuffer<float> chb(kNumFrames, kMono); RTC_EXPECT_DEATH(chb.set_num_channels(kStereo), "num_channels"); } -TEST(IFChannelBufferTest, SetNumChannelsDeathTest) { +TEST(IFChannelBufferDeathTest, SetNumChannelsDeathTest) { IFChannelBuffer ifchb(kNumFrames, kMono); RTC_EXPECT_DEATH(ifchb.ibuf()->set_num_channels(kStereo), "num_channels"); } diff --git a/chromium/third_party/webrtc/common_audio/mocks/mock_smoothing_filter.h b/chromium/third_party/webrtc/common_audio/mocks/mock_smoothing_filter.h index 712049fa6a3..9df49dd11a6 100644 --- a/chromium/third_party/webrtc/common_audio/mocks/mock_smoothing_filter.h +++ b/chromium/third_party/webrtc/common_audio/mocks/mock_smoothing_filter.h @@ -18,9 +18,9 @@ namespace webrtc { class MockSmoothingFilter : public SmoothingFilter { public: - MOCK_METHOD1(AddSample, void(float)); - MOCK_METHOD0(GetAverage, absl::optional<float>()); - MOCK_METHOD1(SetTimeConstantMs, bool(int)); + MOCK_METHOD(void, AddSample, (float), (override)); + MOCK_METHOD(absl::optional<float>, GetAverage, (), (override)); + MOCK_METHOD(bool, SetTimeConstantMs, (int), (override)); }; } // namespace webrtc diff --git a/chromium/third_party/webrtc/common_audio/resampler/push_resampler_unittest.cc b/chromium/third_party/webrtc/common_audio/resampler/push_resampler_unittest.cc index 61b9725b3aa..4724833fbb1 100644 --- a/chromium/third_party/webrtc/common_audio/resampler/push_resampler_unittest.cc +++ b/chromium/third_party/webrtc/common_audio/resampler/push_resampler_unittest.cc @@ -31,19 +31,19 @@ TEST(PushResamplerTest, VerifiesInputParameters) { } #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) -TEST(PushResamplerTest, VerifiesBadInputParameters1) { +TEST(PushResamplerDeathTest, VerifiesBadInputParameters1) { PushResampler<int16_t> resampler; RTC_EXPECT_DEATH(resampler.InitializeIfNeeded(-1, 16000, 1), "src_sample_rate_hz"); } -TEST(PushResamplerTest, VerifiesBadInputParameters2) { +TEST(PushResamplerDeathTest, VerifiesBadInputParameters2) { PushResampler<int16_t> resampler; RTC_EXPECT_DEATH(resampler.InitializeIfNeeded(16000, -1, 1), "dst_sample_rate_hz"); } -TEST(PushResamplerTest, VerifiesBadInputParameters3) { +TEST(PushResamplerDeathTest, VerifiesBadInputParameters3) { PushResampler<int16_t> resampler; RTC_EXPECT_DEATH(resampler.InitializeIfNeeded(16000, 16000, 0), "num_channels"); diff --git a/chromium/third_party/webrtc/common_audio/resampler/sinc_resampler_unittest.cc b/chromium/third_party/webrtc/common_audio/resampler/sinc_resampler_unittest.cc index 7bcd7f146ec..b067b23b880 100644 --- a/chromium/third_party/webrtc/common_audio/resampler/sinc_resampler_unittest.cc +++ b/chromium/third_party/webrtc/common_audio/resampler/sinc_resampler_unittest.cc @@ -40,7 +40,7 @@ static const double kKernelInterpolationFactor = 0.5; // Helper class to ensure ChunkedResample() functions properly. class MockSource : public SincResamplerCallback { public: - MOCK_METHOD2(Run, void(size_t frames, float* destination)); + MOCK_METHOD(void, Run, (size_t frames, float* destination), (override)); }; ACTION(ClearBuffer) { diff --git a/chromium/third_party/webrtc/common_audio/third_party/ooura/fft_size_128/ooura_fft.cc b/chromium/third_party/webrtc/common_audio/third_party/ooura/fft_size_128/ooura_fft.cc index 2918374bbac..6b6d6f1fd79 100644 --- a/chromium/third_party/webrtc/common_audio/third_party/ooura/fft_size_128/ooura_fft.cc +++ b/chromium/third_party/webrtc/common_audio/third_party/ooura/fft_size_128/ooura_fft.cc @@ -313,6 +313,14 @@ static void rftbsub_128_C(float* a) { } // namespace +OouraFft::OouraFft(bool sse2_available) { +#if defined(WEBRTC_ARCH_X86_FAMILY) + use_sse2_ = sse2_available; +#else + use_sse2_ = false; +#endif +} + OouraFft::OouraFft() { #if defined(WEBRTC_ARCH_X86_FAMILY) use_sse2_ = (WebRtc_GetCPUInfo(kSSE2) != 0); diff --git a/chromium/third_party/webrtc/common_audio/third_party/ooura/fft_size_128/ooura_fft.h b/chromium/third_party/webrtc/common_audio/third_party/ooura/fft_size_128/ooura_fft.h index 0cdd6aa66f4..8273dfe58ee 100644 --- a/chromium/third_party/webrtc/common_audio/third_party/ooura/fft_size_128/ooura_fft.h +++ b/chromium/third_party/webrtc/common_audio/third_party/ooura/fft_size_128/ooura_fft.h @@ -38,6 +38,10 @@ void rftbsub_128_neon(float* a); class OouraFft { public: + // Ctor allowing the availability of SSE2 support to be specified. + explicit OouraFft(bool sse2_available); + + // Deprecated: This Ctor will soon be removed. OouraFft(); ~OouraFft(); void Fft(float* a) const; diff --git a/chromium/third_party/webrtc/common_audio/vad/mock/mock_vad.h b/chromium/third_party/webrtc/common_audio/vad/mock/mock_vad.h index afe80ef5e14..5a554ce1f92 100644 --- a/chromium/third_party/webrtc/common_audio/vad/mock/mock_vad.h +++ b/chromium/third_party/webrtc/common_audio/vad/mock/mock_vad.h @@ -18,14 +18,14 @@ namespace webrtc { class MockVad : public Vad { public: - virtual ~MockVad() { Die(); } - MOCK_METHOD0(Die, void()); + ~MockVad() override { Die(); } + MOCK_METHOD(void, Die, ()); - MOCK_METHOD3(VoiceActivity, - enum Activity(const int16_t* audio, - size_t num_samples, - int sample_rate_hz)); - MOCK_METHOD0(Reset, void()); + MOCK_METHOD(enum Activity, + VoiceActivity, + (const int16_t* audio, size_t num_samples, int sample_rate_hz), + (override)); + MOCK_METHOD(void, Reset, (), (override)); }; } // namespace webrtc |