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 <kwiberg@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#19630}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 890988c9cb0f3ccb9517c0f0a6432598dffe54bbdiff --git a/modules/audio_processing/aec/aec_core.cc b/modules/audio_processing/aec/aec_core.cc
index 0eec22f..d580690 100644
--- a/modules/audio_processing/aec/aec_core.cc
+++ b/modules/audio_processing/aec/aec_core.cc
@@ -836,8 +836,6 @@
// 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/modules/audio_processing/aec/aec_resampler.cc b/modules/audio_processing/aec/aec_resampler.cc
index 174fd20..ff2960b 100644
--- a/modules/audio_processing/aec/aec_resampler.cc
+++ b/modules/audio_processing/aec/aec_resampler.cc
@@ -37,7 +37,7 @@
static int EstimateSkew(const int* rawSkew,
int size,
- int absLimit,
+ int deviceSampleRateHz,
float* skewEst);
void* WebRtcAec_CreateResampler() {
diff --git a/modules/audio_processing/aec/echo_cancellation.cc b/modules/audio_processing/aec/echo_cancellation.cc
index 9261632..9e9e707 100644
--- a/modules/audio_processing/aec/echo_cancellation.cc
+++ b/modules/audio_processing/aec/echo_cancellation.cc
@@ -105,15 +105,15 @@
// (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 @@
return reinterpret_cast<Aec*>(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 @@
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 @@
// 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 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 @@
// 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<int16_t>(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 @@
// 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<int16_t>(0.95 * self->filtDelay + 0.05 * current_delay));
+ aecInst->filtDelay = WEBRTC_SPL_MAX(
+ 0, static_cast<int16_t>(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/modules/audio_processing/aec3/echo_canceller3.cc b/modules/audio_processing/aec3/echo_canceller3.cc
index 95dfe8a..143f1b9 100644
--- a/modules/audio_processing/aec3/echo_canceller3.cc
+++ b/modules/audio_processing/aec3/echo_canceller3.cc
@@ -146,7 +146,7 @@
int frame_length,
int num_bands);
~RenderWriter();
- void Insert(AudioBuffer* render);
+ void Insert(AudioBuffer* input);
private:
ApmDataDumper* data_dumper_;
diff --git a/modules/audio_processing/aec3/echo_canceller3_unittest.cc b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
index 3a45500..ac3f709 100644
--- a/modules/audio_processing/aec3/echo_canceller3_unittest.cc
+++ b/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/modules/audio_processing/aec3/echo_remover.cc b/modules/audio_processing/aec3/echo_remover.cc
index 32ecc73..6380a96 100644
--- a/modules/audio_processing/aec3/echo_remover.cc
+++ b/modules/audio_processing/aec3/echo_remover.cc
@@ -57,12 +57,11 @@
// 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<size_t>& external_echo_path_delay_estimate,
- const EchoPathVariability& echo_path_variability,
- bool capture_signal_saturation,
- const RenderBuffer& render_buffer,
- std::vector<std::vector<float>>* capture) override;
+ void ProcessCapture(const rtc::Optional<size_t>& echo_path_delay_samples,
+ const EchoPathVariability& echo_path_variability,
+ bool capture_signal_saturation,
+ const RenderBuffer& render_buffer,
+ std::vector<std::vector<float>>* capture) override;
// Updates the status on whether echo leakage is detected in the output of the
// echo remover.
diff --git a/modules/audio_processing/aecm/echo_control_mobile.cc b/modules/audio_processing/aecm/echo_control_mobile.cc
index a81466e..027ed14 100644
--- a/modules/audio_processing/aecm/echo_control_mobile.cc
+++ b/modules/audio_processing/aecm/echo_control_mobile.cc
@@ -75,10 +75,10 @@
// 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<AecMobile*>(malloc(sizeof(AecMobile)));
diff --git a/modules/audio_processing/agc/agc_manager_direct_unittest.cc b/modules/audio_processing/agc/agc_manager_direct_unittest.cc
index b582ae5..141e99c 100644
--- a/modules/audio_processing/agc/agc_manager_direct_unittest.cc
+++ b/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/modules/audio_processing/audio_processing_impl_unittest.cc b/modules/audio_processing/audio_processing_impl_unittest.cc
index 2ee1d51..75e5aab 100644
--- a/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/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/modules/audio_processing/audio_processing_performance_unittest.cc b/modules/audio_processing/audio_processing_performance_unittest.cc
index 2c20c0f..6d0a61b 100644
--- a/modules/audio_processing/audio_processing_performance_unittest.cc
+++ b/modules/audio_processing/audio_processing_performance_unittest.cc
@@ -258,7 +258,7 @@
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/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 243a140..de070f3 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -237,7 +237,7 @@
}
#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 @@
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 @@
// have competing filenames.
std::map<std::string, std::string> 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 @@
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 @@
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<ExperimentalAgc>(new ExperimentalAgc(false));
std::unique_ptr<AudioProcessing> ap(AudioProcessing::Create(config));
diff --git a/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc b/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc
index 10be164..7acdfc6 100644
--- a/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc
+++ b/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc
@@ -67,7 +67,7 @@
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/modules/audio_processing/residual_echo_detector.cc b/modules/audio_processing/residual_echo_detector.cc
index 67bf4a1..290af43 100644
--- a/modules/audio_processing/residual_echo_detector.cc
+++ b/modules/audio_processing/residual_echo_detector.cc
@@ -22,7 +22,7 @@
namespace {
float Power(rtc::ArrayView<const float> 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/modules/audio_processing/residual_echo_detector_complexity_unittest.cc b/modules/audio_processing/residual_echo_detector_complexity_unittest.cc
index 57a465a..a239279 100644
--- a/modules/audio_processing/residual_echo_detector_complexity_unittest.cc
+++ b/modules/audio_processing/residual_echo_detector_complexity_unittest.cc
@@ -89,7 +89,7 @@
"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/modules/audio_processing/test/aec_dump_based_simulator.cc b/modules/audio_processing/test/aec_dump_based_simulator.cc
index c2e262a..40df9aa 100644
--- a/modules/audio_processing/test/aec_dump_based_simulator.cc
+++ b/modules/audio_processing/test/aec_dump_based_simulator.cc
@@ -482,7 +482,7 @@
}
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/modules/audio_processing/test/audio_processing_simulator.cc b/modules/audio_processing/test/audio_processing_simulator.cc
index 461fc71..3360a67 100644
--- a/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/modules/audio_processing/test/audio_processing_simulator.cc
@@ -82,7 +82,7 @@
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/modules/audio_processing/test/audioproc_float.cc b/modules/audio_processing/test/audioproc_float.cc
index bdf49d7..d0c813f 100644
--- a/modules/audio_processing/test/audioproc_float.cc
+++ b/modules/audio_processing/test/audioproc_float.cc
@@ -175,7 +175,7 @@
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<std::string>* parameter) {
if (value.compare("") != 0) {
*parameter = rtc::Optional<std::string>(value);
@@ -279,7 +279,7 @@
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/modules/audio_processing/test/debug_dump_test.cc b/modules/audio_processing/test/debug_dump_test.cc
index acaadf3..d496b75 100644
--- a/modules/audio_processing/test/debug_dump_test.cc
+++ b/modules/audio_processing/test/debug_dump_test.cc
@@ -40,10 +40,10 @@
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 @@
// 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/modules/audio_processing/test/echo_canceller_test_tools.cc b/modules/audio_processing/test/echo_canceller_test_tools.cc
index 9593da4..b3cacf8 100644
--- a/modules/audio_processing/test/echo_canceller_test_tools.cc
+++ b/modules/audio_processing/test/echo_canceller_test_tools.cc
@@ -24,7 +24,7 @@
void DelayBuffer<T>::Delay(rtc::ArrayView<const T> x,
rtc::ArrayView<T> 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) {
