From 890988c9cb0f3ccb9517c0f0a6432598dffe54bb Mon Sep 17 00:00:00 2001 From: Alex Loiko Date: Thu, 31 Aug 2017 10:25:48 +0200 Subject: [PATCH] Run the ClangTidy analyser on the AudioProcessing submodule of WebRTC. This CL contains automatically applied fixes suggested by the ClangTidy analyzer (http://clang.llvm.org/extra/clang-tidy/). The following kinds of fixes is present: * renaming variables when the names in the method signature don't match the names in the method definition (ClangTidy:readability-inconsistent-declaration-parameter-name) * ClangTidy:readability-container-size-empty, ClangTidy:misc-unused-using-decls, ClangTidy:performance-unnecessary-value-param, ClangTidy:readability-redundant-control-flow This is a 'pilot' CL to check if automatic code analyzers can feasibly be integrated into the WebRTC infrastructuve. The renamings have been manually expected for consistency with surrounding code. In echo_cancellation.cc, I changed several names in the function implementation to match the function declaration. The tool suggested changing everything to match the function definitions instead. Bug: None Change-Id: Id3b7ba18c51f15b025f26090c7bdcc642e48d8fd Reviewed-on: https://chromium-review.googlesource.com/635766 Reviewed-by: Karl Wiberg Commit-Queue: Alex Loiko Cr-Commit-Position: refs/heads/master@{#19630} --- .../modules/audio_processing/aec/aec_core.cc | 2 - .../audio_processing/aec/aec_resampler.cc | 2 +- .../audio_processing/aec/echo_cancellation.cc | 203 +++++++++--------- .../audio_processing/aec3/echo_canceller3.cc | 2 +- .../aec3/echo_canceller3_unittest.cc | 1 - .../audio_processing/aec3/echo_remover.cc | 11 +- .../aecm/echo_control_mobile.cc | 4 +- .../agc/agc_manager_direct_unittest.cc | 3 - .../audio_processing_impl_unittest.cc | 1 - .../audio_processing_performance_unittest.cc | 2 +- .../audio_processing_unittest.cc | 10 +- .../level_controller_complexity_unittest.cc | 2 +- .../residual_echo_detector.cc | 2 +- ...idual_echo_detector_complexity_unittest.cc | 2 +- .../test/aec_dump_based_simulator.cc | 2 +- .../test/audio_processing_simulator.cc | 2 +- .../audio_processing/test/audioproc_float.cc | 4 +- .../audio_processing/test/debug_dump_test.cc | 6 +- .../test/echo_canceller_test_tools.cc | 2 +- 19 files changed, 130 insertions(+), 133 deletions(-) diff --git a/webrtc/modules/audio_processing/aec/aec_core.cc b/webrtc/modules/audio_processing/aec/aec_core.cc index 0eec22f4e5..d580690804 100644 --- a/webrtc/modules/audio_processing/aec/aec_core.cc +++ b/webrtc/modules/audio_processing/aec/aec_core.cc @@ -836,8 +836,6 @@ static void UpdateDelayMetrics(AecCore* self) { // Reset histogram. memset(self->delay_histogram, 0, sizeof(self->delay_histogram)); self->num_delay_values = 0; - - return; } static void ScaledInverseFft(const OouraFft& ooura_fft, diff --git a/webrtc/modules/audio_processing/aec/aec_resampler.cc b/webrtc/modules/audio_processing/aec/aec_resampler.cc index 174fd20c34..ff2960bd30 100644 --- a/webrtc/modules/audio_processing/aec/aec_resampler.cc +++ b/webrtc/modules/audio_processing/aec/aec_resampler.cc @@ -37,7 +37,7 @@ typedef struct { static int EstimateSkew(const int* rawSkew, int size, - int absLimit, + int deviceSampleRateHz, float* skewEst); void* WebRtcAec_CreateResampler() { diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation.cc b/webrtc/modules/audio_processing/aec/echo_cancellation.cc index 9261632c17..9e9e7076ce 100644 --- a/webrtc/modules/audio_processing/aec/echo_cancellation.cc +++ b/webrtc/modules/audio_processing/aec/echo_cancellation.cc @@ -105,15 +105,15 @@ int Aec::instance_count = 0; // (controlled by knownDelay) static void EstBufDelayNormal(Aec* aecInst); static void EstBufDelayExtended(Aec* aecInst); -static int ProcessNormal(Aec* self, - const float* const* near, +static int ProcessNormal(Aec* aecInst, + const float* const* nearend, size_t num_bands, float* const* out, size_t num_samples, int16_t reported_delay_ms, int32_t skew); -static void ProcessExtended(Aec* self, - const float* const* near, +static void ProcessExtended(Aec* aecInst, + const float* const* nearend, size_t num_bands, float* const* out, size_t num_samples, @@ -531,12 +531,12 @@ AecCore* WebRtcAec_aec_core(void* handle) { return reinterpret_cast(handle)->aec; } -static int ProcessNormal(Aec* aecpc, +static int ProcessNormal(Aec* aecInst, const float* const* nearend, size_t num_bands, float* const* out, - size_t nrOfSamples, - int16_t msInSndCardBuf, + size_t num_samples, + int16_t reported_delay_ms, int32_t skew) { int retVal = 0; size_t i; @@ -545,47 +545,48 @@ static int ProcessNormal(Aec* aecpc, const float minSkewEst = -0.5f; const float maxSkewEst = 1.0f; - msInSndCardBuf = - msInSndCardBuf > kMaxTrustedDelayMs ? kMaxTrustedDelayMs : msInSndCardBuf; + reported_delay_ms = + reported_delay_ms > kMaxTrustedDelayMs ? kMaxTrustedDelayMs : + reported_delay_ms; // TODO(andrew): we need to investigate if this +10 is really wanted. - msInSndCardBuf += 10; - aecpc->msInSndCardBuf = msInSndCardBuf; + reported_delay_ms += 10; + aecInst->msInSndCardBuf = reported_delay_ms; - if (aecpc->skewMode == kAecTrue) { - if (aecpc->skewFrCtr < 25) { - aecpc->skewFrCtr++; + if (aecInst->skewMode == kAecTrue) { + if (aecInst->skewFrCtr < 25) { + aecInst->skewFrCtr++; } else { - retVal = WebRtcAec_GetSkew(aecpc->resampler, skew, &aecpc->skew); + retVal = WebRtcAec_GetSkew(aecInst->resampler, skew, &aecInst->skew); if (retVal == -1) { - aecpc->skew = 0; + aecInst->skew = 0; retVal = AEC_BAD_PARAMETER_WARNING; } - aecpc->skew /= aecpc->sampFactor * nrOfSamples; + aecInst->skew /= aecInst->sampFactor * num_samples; - if (aecpc->skew < 1.0e-3 && aecpc->skew > -1.0e-3) { - aecpc->resample = kAecFalse; + if (aecInst->skew < 1.0e-3 && aecInst->skew > -1.0e-3) { + aecInst->resample = kAecFalse; } else { - aecpc->resample = kAecTrue; + aecInst->resample = kAecTrue; } - if (aecpc->skew < minSkewEst) { - aecpc->skew = minSkewEst; - } else if (aecpc->skew > maxSkewEst) { - aecpc->skew = maxSkewEst; + if (aecInst->skew < minSkewEst) { + aecInst->skew = minSkewEst; + } else if (aecInst->skew > maxSkewEst) { + aecInst->skew = maxSkewEst; } - aecpc->data_dumper->DumpRaw("aec_skew", 1, &aecpc->skew); + aecInst->data_dumper->DumpRaw("aec_skew", 1, &aecInst->skew); } } - nBlocks10ms = nrOfSamples / (FRAME_LEN * aecpc->rate_factor); + nBlocks10ms = num_samples / (FRAME_LEN * aecInst->rate_factor); - if (aecpc->startup_phase) { + if (aecInst->startup_phase) { for (i = 0; i < num_bands; ++i) { // Only needed if they don't already point to the same place. if (nearend[i] != out[i]) { - memcpy(out[i], nearend[i], sizeof(nearend[i][0]) * nrOfSamples); + memcpy(out[i], nearend[i], sizeof(nearend[i][0]) * num_samples); } } @@ -593,82 +594,83 @@ static int ProcessNormal(Aec* aecpc, // AEC is disabled until the system delay is OK // Mechanism to ensure that the system delay is reasonably stable. - if (aecpc->checkBuffSize) { - aecpc->checkBufSizeCtr++; + if (aecInst->checkBuffSize) { + aecInst->checkBufSizeCtr++; // Before we fill up the far-end buffer we require the system delay // to be stable (+/-8 ms) compared to the first value. This // comparison is made during the following 6 consecutive 10 ms // blocks. If it seems to be stable then we start to fill up the // far-end buffer. - if (aecpc->counter == 0) { - aecpc->firstVal = aecpc->msInSndCardBuf; - aecpc->sum = 0; + if (aecInst->counter == 0) { + aecInst->firstVal = aecInst->msInSndCardBuf; + aecInst->sum = 0; } - if (abs(aecpc->firstVal - aecpc->msInSndCardBuf) < - WEBRTC_SPL_MAX(0.2 * aecpc->msInSndCardBuf, sampMsNb)) { - aecpc->sum += aecpc->msInSndCardBuf; - aecpc->counter++; + if (abs(aecInst->firstVal - aecInst->msInSndCardBuf) < + WEBRTC_SPL_MAX(0.2 * aecInst->msInSndCardBuf, sampMsNb)) { + aecInst->sum += aecInst->msInSndCardBuf; + aecInst->counter++; } else { - aecpc->counter = 0; + aecInst->counter = 0; } - if (aecpc->counter * nBlocks10ms >= 6) { + if (aecInst->counter * nBlocks10ms >= 6) { // The far-end buffer size is determined in partitions of // PART_LEN samples. Use 75% of the average value of the system // delay as buffer size to start with. - aecpc->bufSizeStart = - WEBRTC_SPL_MIN((3 * aecpc->sum * aecpc->rate_factor * 8) / - (4 * aecpc->counter * PART_LEN), + aecInst->bufSizeStart = + WEBRTC_SPL_MIN((3 * aecInst->sum * aecInst->rate_factor * 8) / + (4 * aecInst->counter * PART_LEN), kMaxBufSizeStart); // Buffer size has now been determined. - aecpc->checkBuffSize = 0; + aecInst->checkBuffSize = 0; } - if (aecpc->checkBufSizeCtr * nBlocks10ms > 50) { + if (aecInst->checkBufSizeCtr * nBlocks10ms > 50) { // For really bad systems, don't disable the echo canceller for // more than 0.5 sec. - aecpc->bufSizeStart = WEBRTC_SPL_MIN( - (aecpc->msInSndCardBuf * aecpc->rate_factor * 3) / 40, + aecInst->bufSizeStart = WEBRTC_SPL_MIN( + (aecInst->msInSndCardBuf * aecInst->rate_factor * 3) / 40, kMaxBufSizeStart); - aecpc->checkBuffSize = 0; + aecInst->checkBuffSize = 0; } } // If |checkBuffSize| changed in the if-statement above. - if (!aecpc->checkBuffSize) { + if (!aecInst->checkBuffSize) { // The system delay is now reasonably stable (or has been unstable // for too long). When the far-end buffer is filled with // approximately the same amount of data as reported by the system // we end the startup phase. int overhead_elements = - WebRtcAec_system_delay(aecpc->aec) / PART_LEN - aecpc->bufSizeStart; + WebRtcAec_system_delay(aecInst->aec) / PART_LEN - + aecInst->bufSizeStart; if (overhead_elements == 0) { // Enable the AEC - aecpc->startup_phase = 0; + aecInst->startup_phase = 0; } else if (overhead_elements > 0) { // TODO(bjornv): Do we need a check on how much we actually // moved the read pointer? It should always be possible to move // the pointer |overhead_elements| since we have only added data // to the buffer and no delay compensation nor AEC processing // has been done. - WebRtcAec_AdjustFarendBufferSizeAndSystemDelay(aecpc->aec, + WebRtcAec_AdjustFarendBufferSizeAndSystemDelay(aecInst->aec, overhead_elements); // Enable the AEC - aecpc->startup_phase = 0; + aecInst->startup_phase = 0; } } } else { // AEC is enabled. - EstBufDelayNormal(aecpc); + EstBufDelayNormal(aecInst); // Call the AEC. // TODO(bjornv): Re-structure such that we don't have to pass - // |aecpc->knownDelay| as input. Change name to something like + // |aecInst->knownDelay| as input. Change name to something like // |system_buffer_diff|. - WebRtcAec_ProcessFrames(aecpc->aec, nearend, num_bands, nrOfSamples, - aecpc->knownDelay, out); + WebRtcAec_ProcessFrames(aecInst->aec, nearend, num_bands, num_samples, + aecInst->knownDelay, out); } return retVal; @@ -749,9 +751,9 @@ static void ProcessExtended(Aec* self, } } -static void EstBufDelayNormal(Aec* aecpc) { - int nSampSndCard = aecpc->msInSndCardBuf * sampMsNb * aecpc->rate_factor; - int current_delay = nSampSndCard - WebRtcAec_system_delay(aecpc->aec); +static void EstBufDelayNormal(Aec* aecInst) { + int nSampSndCard = aecInst->msInSndCardBuf * sampMsNb * aecInst->rate_factor; + int current_delay = nSampSndCard - WebRtcAec_system_delay(aecInst->aec); int delay_difference = 0; // Before we proceed with the delay estimate filtering we: @@ -761,54 +763,55 @@ static void EstBufDelayNormal(Aec* aecpc) { // be negative. // 1) Compensating for the frame(s) that will be read/processed. - current_delay += FRAME_LEN * aecpc->rate_factor; + current_delay += FRAME_LEN * aecInst->rate_factor; // 2) Account for resampling frame delay. - if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) { + if (aecInst->skewMode == kAecTrue && aecInst->resample == kAecTrue) { current_delay -= kResamplingDelay; } // 3) Compensate for non-causality, if needed, by flushing one block. if (current_delay < PART_LEN) { current_delay += - WebRtcAec_AdjustFarendBufferSizeAndSystemDelay(aecpc->aec, 1) * + WebRtcAec_AdjustFarendBufferSizeAndSystemDelay(aecInst->aec, 1) * PART_LEN; } // We use -1 to signal an initialized state in the "extended" implementation; // compensate for that. - aecpc->filtDelay = aecpc->filtDelay < 0 ? 0 : aecpc->filtDelay; - aecpc->filtDelay = + aecInst->filtDelay = aecInst->filtDelay < 0 ? 0 : aecInst->filtDelay; + aecInst->filtDelay = WEBRTC_SPL_MAX(0, static_cast(0.8 * - aecpc->filtDelay + + aecInst->filtDelay + 0.2 * current_delay)); - delay_difference = aecpc->filtDelay - aecpc->knownDelay; + delay_difference = aecInst->filtDelay - aecInst->knownDelay; if (delay_difference > 224) { - if (aecpc->lastDelayDiff < 96) { - aecpc->timeForDelayChange = 0; + if (aecInst->lastDelayDiff < 96) { + aecInst->timeForDelayChange = 0; } else { - aecpc->timeForDelayChange++; + aecInst->timeForDelayChange++; } - } else if (delay_difference < 96 && aecpc->knownDelay > 0) { - if (aecpc->lastDelayDiff > 224) { - aecpc->timeForDelayChange = 0; + } else if (delay_difference < 96 && aecInst->knownDelay > 0) { + if (aecInst->lastDelayDiff > 224) { + aecInst->timeForDelayChange = 0; } else { - aecpc->timeForDelayChange++; + aecInst->timeForDelayChange++; } } else { - aecpc->timeForDelayChange = 0; + aecInst->timeForDelayChange = 0; } - aecpc->lastDelayDiff = delay_difference; + aecInst->lastDelayDiff = delay_difference; - if (aecpc->timeForDelayChange > 25) { - aecpc->knownDelay = WEBRTC_SPL_MAX((int)aecpc->filtDelay - 160, 0); + if (aecInst->timeForDelayChange > 25) { + aecInst->knownDelay = WEBRTC_SPL_MAX((int)aecInst->filtDelay - 160, 0); } } -static void EstBufDelayExtended(Aec* self) { - int reported_delay = self->msInSndCardBuf * sampMsNb * self->rate_factor; - int current_delay = reported_delay - WebRtcAec_system_delay(self->aec); +static void EstBufDelayExtended(Aec* aecInst) { + int reported_delay = aecInst->msInSndCardBuf * sampMsNb * + aecInst->rate_factor; + int current_delay = reported_delay - WebRtcAec_system_delay(aecInst->aec); int delay_difference = 0; // Before we proceed with the delay estimate filtering we: @@ -818,46 +821,48 @@ static void EstBufDelayExtended(Aec* self) { // be negative. // 1) Compensating for the frame(s) that will be read/processed. - current_delay += FRAME_LEN * self->rate_factor; + current_delay += FRAME_LEN * aecInst->rate_factor; // 2) Account for resampling frame delay. - if (self->skewMode == kAecTrue && self->resample == kAecTrue) { + if (aecInst->skewMode == kAecTrue && aecInst->resample == kAecTrue) { current_delay -= kResamplingDelay; } // 3) Compensate for non-causality, if needed, by flushing two blocks. if (current_delay < PART_LEN) { current_delay += - WebRtcAec_AdjustFarendBufferSizeAndSystemDelay(self->aec, 2) * PART_LEN; + WebRtcAec_AdjustFarendBufferSizeAndSystemDelay(aecInst->aec, 2) * + PART_LEN; } - if (self->filtDelay == -1) { - self->filtDelay = WEBRTC_SPL_MAX(0, 0.5 * current_delay); + if (aecInst->filtDelay == -1) { + aecInst->filtDelay = WEBRTC_SPL_MAX(0, 0.5 * current_delay); } else { - self->filtDelay = WEBRTC_SPL_MAX( - 0, static_cast(0.95 * self->filtDelay + 0.05 * current_delay)); + aecInst->filtDelay = WEBRTC_SPL_MAX( + 0, static_cast(0.95 * aecInst->filtDelay + 0.05 * + current_delay)); } - delay_difference = self->filtDelay - self->knownDelay; + delay_difference = aecInst->filtDelay - aecInst->knownDelay; if (delay_difference > 384) { - if (self->lastDelayDiff < 128) { - self->timeForDelayChange = 0; + if (aecInst->lastDelayDiff < 128) { + aecInst->timeForDelayChange = 0; } else { - self->timeForDelayChange++; + aecInst->timeForDelayChange++; } - } else if (delay_difference < 128 && self->knownDelay > 0) { - if (self->lastDelayDiff > 384) { - self->timeForDelayChange = 0; + } else if (delay_difference < 128 && aecInst->knownDelay > 0) { + if (aecInst->lastDelayDiff > 384) { + aecInst->timeForDelayChange = 0; } else { - self->timeForDelayChange++; + aecInst->timeForDelayChange++; } } else { - self->timeForDelayChange = 0; + aecInst->timeForDelayChange = 0; } - self->lastDelayDiff = delay_difference; + aecInst->lastDelayDiff = delay_difference; - if (self->timeForDelayChange > 25) { - self->knownDelay = WEBRTC_SPL_MAX((int)self->filtDelay - 256, 0); + if (aecInst->timeForDelayChange > 25) { + aecInst->knownDelay = WEBRTC_SPL_MAX((int)aecInst->filtDelay - 256, 0); } } } // namespace webrtc diff --git a/webrtc/modules/audio_processing/aec3/echo_canceller3.cc b/webrtc/modules/audio_processing/aec3/echo_canceller3.cc index 95dfe8a743..143f1b9d2a 100644 --- a/webrtc/modules/audio_processing/aec3/echo_canceller3.cc +++ b/webrtc/modules/audio_processing/aec3/echo_canceller3.cc @@ -146,7 +146,7 @@ class EchoCanceller3::RenderWriter { int frame_length, int num_bands); ~RenderWriter(); - void Insert(AudioBuffer* render); + void Insert(AudioBuffer* input); private: ApmDataDumper* data_dumper_; diff --git a/webrtc/modules/audio_processing/aec3/echo_canceller3_unittest.cc b/webrtc/modules/audio_processing/aec3/echo_canceller3_unittest.cc index 3a4550077d..ac3f7091b3 100644 --- a/webrtc/modules/audio_processing/aec3/echo_canceller3_unittest.cc +++ b/webrtc/modules/audio_processing/aec3/echo_canceller3_unittest.cc @@ -28,7 +28,6 @@ namespace webrtc { namespace { -using testing::Return; using testing::StrictMock; using testing::_; diff --git a/webrtc/modules/audio_processing/aec3/echo_remover.cc b/webrtc/modules/audio_processing/aec3/echo_remover.cc index 32ecc73926..6380a96820 100644 --- a/webrtc/modules/audio_processing/aec3/echo_remover.cc +++ b/webrtc/modules/audio_processing/aec3/echo_remover.cc @@ -57,12 +57,11 @@ class EchoRemoverImpl final : public EchoRemover { // Removes the echo from a block of samples from the capture signal. The // supplied render signal is assumed to be pre-aligned with the capture // signal. - void ProcessCapture( - const rtc::Optional& external_echo_path_delay_estimate, - const EchoPathVariability& echo_path_variability, - bool capture_signal_saturation, - const RenderBuffer& render_buffer, - std::vector>* capture) override; + void ProcessCapture(const rtc::Optional& echo_path_delay_samples, + const EchoPathVariability& echo_path_variability, + bool capture_signal_saturation, + const RenderBuffer& render_buffer, + std::vector>* capture) override; // Updates the status on whether echo leakage is detected in the output of the // echo remover. diff --git a/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc b/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc index a81466e678..027ed14cf7 100644 --- a/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc +++ b/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc @@ -75,10 +75,10 @@ typedef struct // Estimates delay to set the position of the farend buffer read pointer // (controlled by knownDelay) -static int WebRtcAecm_EstBufDelay(AecMobile* aecmInst, short msInSndCardBuf); +static int WebRtcAecm_EstBufDelay(AecMobile* aecm, short msInSndCardBuf); // Stuffs the farend buffer if the estimated delay is too large -static int WebRtcAecm_DelayComp(AecMobile* aecmInst); +static int WebRtcAecm_DelayComp(AecMobile* aecm); void* WebRtcAecm_Create() { AecMobile* aecm = static_cast(malloc(sizeof(AecMobile))); diff --git a/webrtc/modules/audio_processing/agc/agc_manager_direct_unittest.cc b/webrtc/modules/audio_processing/agc/agc_manager_direct_unittest.cc index b582ae5669..141e99cd62 100644 --- a/webrtc/modules/audio_processing/agc/agc_manager_direct_unittest.cc +++ b/webrtc/modules/audio_processing/agc/agc_manager_direct_unittest.cc @@ -20,11 +20,8 @@ using ::testing::_; using ::testing::DoAll; -using ::testing::Eq; -using ::testing::Mock; using ::testing::Return; using ::testing::SetArgPointee; -using ::testing::SetArgReferee; namespace webrtc { namespace { diff --git a/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc b/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc index 2ee1d51936..75e5aab60e 100644 --- a/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc +++ b/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc @@ -17,7 +17,6 @@ #include "webrtc/test/gtest.h" using ::testing::Invoke; -using ::testing::Return; namespace webrtc { namespace { diff --git a/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc b/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc index 2c20c0fb72..6d0a61b07a 100644 --- a/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc +++ b/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc @@ -258,7 +258,7 @@ class TimedThreadApiProcessor { bool Process(); // Method for printing out the simulation statistics. - void print_processor_statistics(std::string processor_name) const { + void print_processor_statistics(const std::string& processor_name) const { const std::string modifier = "_api_call_duration"; // Lambda function for creating a test printout string. diff --git a/webrtc/modules/audio_processing/audio_processing_unittest.cc b/webrtc/modules/audio_processing/audio_processing_unittest.cc index 243a140440..de070f3ef9 100644 --- a/webrtc/modules/audio_processing/audio_processing_unittest.cc +++ b/webrtc/modules/audio_processing/audio_processing_unittest.cc @@ -237,7 +237,7 @@ void WriteStatsMessage(const AudioProcessing::Statistic& output, } #endif -void OpenFileAndWriteMessage(const std::string filename, +void OpenFileAndWriteMessage(const std::string& filename, const MessageLite& msg) { FILE* file = fopen(filename.c_str(), "wb"); ASSERT_TRUE(file != NULL); @@ -253,7 +253,7 @@ void OpenFileAndWriteMessage(const std::string filename, fclose(file); } -std::string ResourceFilePath(std::string name, int sample_rate_hz) { +std::string ResourceFilePath(const std::string& name, int sample_rate_hz) { std::ostringstream ss; // Resource files are all stereo. ss << name << sample_rate_hz / 1000 << "_stereo"; @@ -265,7 +265,7 @@ std::string ResourceFilePath(std::string name, int sample_rate_hz) { // have competing filenames. std::map temp_filenames; -std::string OutputFilePath(std::string name, +std::string OutputFilePath(const std::string& name, int input_rate, int output_rate, int reverse_input_rate, @@ -307,7 +307,7 @@ void ClearTempFiles() { remove(kv.second.c_str()); } -void OpenFileAndReadMessage(std::string filename, MessageLite* msg) { +void OpenFileAndReadMessage(const std::string& filename, MessageLite* msg) { FILE* file = fopen(filename.c_str(), "rb"); ASSERT_TRUE(file != NULL); ReadMessageFromFile(file, msg); @@ -2438,7 +2438,7 @@ class AudioProcessingTest size_t num_output_channels, size_t num_reverse_input_channels, size_t num_reverse_output_channels, - std::string output_file_prefix) { + const std::string& output_file_prefix) { Config config; config.Set(new ExperimentalAgc(false)); std::unique_ptr ap(AudioProcessing::Create(config)); diff --git a/webrtc/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc b/webrtc/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc index 10be164f8b..7acdfc6307 100644 --- a/webrtc/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc +++ b/webrtc/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc @@ -67,7 +67,7 @@ void RunStandaloneSubmodule(int sample_rate_hz, size_t num_channels) { false); } -void RunTogetherWithApm(std::string test_description, +void RunTogetherWithApm(const std::string& test_description, int render_input_sample_rate_hz, int render_output_sample_rate_hz, int capture_input_sample_rate_hz, diff --git a/webrtc/modules/audio_processing/residual_echo_detector.cc b/webrtc/modules/audio_processing/residual_echo_detector.cc index 67bf4a1f99..290af43a5f 100644 --- a/webrtc/modules/audio_processing/residual_echo_detector.cc +++ b/webrtc/modules/audio_processing/residual_echo_detector.cc @@ -22,7 +22,7 @@ namespace { float Power(rtc::ArrayView input) { - if (input.size() == 0) { + if (input.empty()) { return 0.f; } return std::inner_product(input.begin(), input.end(), input.begin(), 0.f) / diff --git a/webrtc/modules/audio_processing/residual_echo_detector_complexity_unittest.cc b/webrtc/modules/audio_processing/residual_echo_detector_complexity_unittest.cc index 57a465a85a..a239279db0 100644 --- a/webrtc/modules/audio_processing/residual_echo_detector_complexity_unittest.cc +++ b/webrtc/modules/audio_processing/residual_echo_detector_complexity_unittest.cc @@ -89,7 +89,7 @@ void RunStandaloneSubmodule() { "us", false); } -void RunTogetherWithApm(std::string test_description, +void RunTogetherWithApm(const std::string& test_description, bool use_mobile_aec, bool include_default_apm_processing) { test::SimulatorBuffers buffers( diff --git a/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc index c2e262ae6f..40df9aa120 100644 --- a/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc +++ b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc @@ -482,7 +482,7 @@ void AecDumpBasedSimulator::HandleMessage( } if (settings_.use_verbose_logging && msg.has_experiments_description() && - msg.experiments_description().size() > 0) { + !msg.experiments_description().empty()) { std::cout << " experiments not included by default in the simulation: " << msg.experiments_description() << std::endl; } diff --git a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc index 461fc71c5e..3360a672ec 100644 --- a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc +++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc @@ -82,7 +82,7 @@ AudioProcessingSimulator::AudioProcessingSimulator( const SimulationSettings& settings) : settings_(settings), worker_queue_("file_writer_task_queue") { if (settings_.ed_graph_output_filename && - settings_.ed_graph_output_filename->size() > 0) { + !settings_.ed_graph_output_filename->empty()) { residual_echo_likelihood_graph_writer_.open( *settings_.ed_graph_output_filename); RTC_CHECK(residual_echo_likelihood_graph_writer_.is_open()); diff --git a/webrtc/modules/audio_processing/test/audioproc_float.cc b/webrtc/modules/audio_processing/test/audioproc_float.cc index bdf49d7afa..d0c813f151 100644 --- a/webrtc/modules/audio_processing/test/audioproc_float.cc +++ b/webrtc/modules/audio_processing/test/audioproc_float.cc @@ -175,7 +175,7 @@ DEFINE_bool(store_intermediate_output, DEFINE_string(custom_call_order_file, "", "Custom process API call order file"); DEFINE_bool(help, false, "Print this message"); -void SetSettingIfSpecified(const std::string value, +void SetSettingIfSpecified(const std::string& value, rtc::Optional* parameter) { if (value.compare("") != 0) { *parameter = rtc::Optional(value); @@ -279,7 +279,7 @@ SimulationSettings CreateSettings() { return settings; } -void ReportConditionalErrorAndExit(bool condition, std::string message) { +void ReportConditionalErrorAndExit(bool condition, const std::string& message) { if (condition) { std::cerr << message << std::endl; exit(1); diff --git a/webrtc/modules/audio_processing/test/debug_dump_test.cc b/webrtc/modules/audio_processing/test/debug_dump_test.cc index acaadf391e..d496b755e8 100644 --- a/webrtc/modules/audio_processing/test/debug_dump_test.cc +++ b/webrtc/modules/audio_processing/test/debug_dump_test.cc @@ -40,10 +40,10 @@ void MaybeResetBuffer(std::unique_ptr>* buffer, class DebugDumpGenerator { public: DebugDumpGenerator(const std::string& input_file_name, - int input_file_rate_hz, + int input_rate_hz, int input_channels, const std::string& reverse_file_name, - int reverse_file_rate_hz, + int reverse_rate_hz, int reverse_channels, const Config& config, const std::string& dump_file_name); @@ -244,7 +244,7 @@ class DebugDumpTest : public ::testing::Test { // VerifyDebugDump replays a debug dump using APM and verifies that the result // is bit-exact-identical to the output channel in the dump. This is only // guaranteed if the debug dump is started on the first frame. - void VerifyDebugDump(const std::string& dump_file_name); + void VerifyDebugDump(const std::string& in_filename); private: DebugDumpReplayer debug_dump_replayer_; diff --git a/webrtc/modules/audio_processing/test/echo_canceller_test_tools.cc b/webrtc/modules/audio_processing/test/echo_canceller_test_tools.cc index 9593da45f0..b3cacf83f7 100644 --- a/webrtc/modules/audio_processing/test/echo_canceller_test_tools.cc +++ b/webrtc/modules/audio_processing/test/echo_canceller_test_tools.cc @@ -24,7 +24,7 @@ template void DelayBuffer::Delay(rtc::ArrayView x, rtc::ArrayView x_delayed) { RTC_DCHECK_EQ(x.size(), x_delayed.size()); - if (buffer_.size() == 0) { + if (buffer_.empty()) { std::copy(x.begin(), x.end(), x_delayed.begin()); } else { for (size_t k = 0; k < x.size(); ++k) {