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Unified Diff: webrtc/modules/audio_processing/test/audio_processing_unittest.cc

Issue 1316523002: Convert channel counts to size_t. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Rebase onto cleanup change Created 5 years ago
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« webrtc/modules/audio_device/include/audio_device_defines.h ('K') | « webrtc/modules/audio_processing/test/audio_file_processor.cc ('k') | webrtc/modules/audio_processing/test/audioproc_float.cc » ('j') | webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h » ('J')
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Index: webrtc/modules/audio_processing/test/audio_processing_unittest.cc
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
index c6818e9c121a59d8fd0f20b619a09eca9afa825b..baab95fbbff7ae736f7c255e4a02eafc1c2ba54a 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -70,7 +70,7 @@ void ConvertToFloat(const int16_t* int_data, ChannelBuffer<float>* cb) {
cb->num_frames(),
cb->num_channels(),
cb_int.channels());
- for (int i = 0; i < cb->num_channels(); ++i) {
+ for (size_t i = 0; i < cb->num_channels(); ++i) {
S16ToFloat(cb_int.channels()[i],
cb->num_frames(),
cb->channels()[i]);
@@ -82,7 +82,7 @@ void ConvertToFloat(const AudioFrame& frame, ChannelBuffer<float>* cb) {
}
// Number of channels including the keyboard channel.
-int TotalChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
+size_t TotalChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
switch (layout) {
case AudioProcessing::kMono:
return 1;
@@ -132,7 +132,7 @@ void SetFrameTo(AudioFrame* frame, int16_t value) {
}
void SetFrameTo(AudioFrame* frame, int16_t left, int16_t right) {
- ASSERT_EQ(2, frame->num_channels_);
+ ASSERT_EQ(2u, frame->num_channels_);
for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
frame->data_[i] = left;
frame->data_[i + 1] = right;
@@ -257,10 +257,10 @@ std::string OutputFilePath(std::string name,
int output_rate,
int reverse_input_rate,
int reverse_output_rate,
- int num_input_channels,
- int num_output_channels,
- int num_reverse_input_channels,
- int num_reverse_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
+ size_t num_reverse_input_channels,
+ size_t num_reverse_output_channels,
StreamDirection file_direction) {
std::ostringstream ss;
ss << name << "_i" << num_input_channels << "_" << input_rate / 1000 << "_ir"
@@ -355,9 +355,9 @@ class ApmTest : public ::testing::Test {
void Init(int sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
- int num_input_channels,
- int num_output_channels,
- int num_reverse_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
+ size_t num_reverse_channels,
bool open_output_file);
void Init(AudioProcessing* ap);
void EnableAllComponents();
@@ -370,12 +370,12 @@ class ApmTest : public ::testing::Test {
void ProcessDelayVerificationTest(int delay_ms, int system_delay_ms,
int delay_min, int delay_max);
void TestChangingChannelsInt16Interface(
- int num_channels,
+ size_t num_channels,
AudioProcessing::Error expected_return);
- void TestChangingForwardChannels(int num_in_channels,
- int num_out_channels,
+ void TestChangingForwardChannels(size_t num_in_channels,
+ size_t num_out_channels,
AudioProcessing::Error expected_return);
- void TestChangingReverseChannels(int num_rev_channels,
+ void TestChangingReverseChannels(size_t num_rev_channels,
AudioProcessing::Error expected_return);
void RunQuantizedVolumeDoesNotGetStuckTest(int sample_rate);
void RunManualVolumeChangeIsPossibleTest(int sample_rate);
@@ -396,7 +396,7 @@ class ApmTest : public ::testing::Test {
rtc::scoped_ptr<ChannelBuffer<float> > float_cb_;
rtc::scoped_ptr<ChannelBuffer<float> > revfloat_cb_;
int output_sample_rate_hz_;
- int num_output_channels_;
+ size_t num_output_channels_;
FILE* far_file_;
FILE* near_file_;
FILE* out_file_;
@@ -480,9 +480,9 @@ void ApmTest::Init(AudioProcessing* ap) {
void ApmTest::Init(int sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
- int num_input_channels,
- int num_output_channels,
- int num_reverse_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
+ size_t num_reverse_channels,
bool open_output_file) {
SetContainerFormat(sample_rate_hz, num_input_channels, frame_, &float_cb_);
output_sample_rate_hz_ = output_sample_rate_hz;
@@ -814,7 +814,7 @@ TEST_F(ApmTest, DelayOffsetWithLimitsIsSetProperly) {
}
void ApmTest::TestChangingChannelsInt16Interface(
- int num_channels,
+ size_t num_channels,
AudioProcessing::Error expected_return) {
frame_->num_channels_ = num_channels;
EXPECT_EQ(expected_return, apm_->ProcessStream(frame_));
@@ -822,8 +822,8 @@ void ApmTest::TestChangingChannelsInt16Interface(
}
void ApmTest::TestChangingForwardChannels(
- int num_in_channels,
- int num_out_channels,
+ size_t num_in_channels,
+ size_t num_out_channels,
AudioProcessing::Error expected_return) {
const StreamConfig input_stream = {frame_->sample_rate_hz_, num_in_channels};
const StreamConfig output_stream = {output_sample_rate_hz_, num_out_channels};
@@ -834,7 +834,7 @@ void ApmTest::TestChangingForwardChannels(
}
void ApmTest::TestChangingReverseChannels(
- int num_rev_channels,
+ size_t num_rev_channels,
AudioProcessing::Error expected_return) {
const ProcessingConfig processing_config = {
{{frame_->sample_rate_hz_, apm_->num_input_channels()},
@@ -855,11 +855,11 @@ TEST_F(ApmTest, ChannelsInt16Interface) {
TestChangingChannelsInt16Interface(0, apm_->kBadNumberChannelsError);
- for (int i = 1; i < 4; i++) {
+ for (size_t i = 1; i < 4; i++) {
TestChangingChannelsInt16Interface(i, kNoErr);
EXPECT_EQ(i, apm_->num_input_channels());
// We always force the number of reverse channels used for processing to 1.
- EXPECT_EQ(1, apm_->num_reverse_channels());
+ EXPECT_EQ(1u, apm_->num_reverse_channels());
}
}
@@ -870,8 +870,8 @@ TEST_F(ApmTest, Channels) {
TestChangingForwardChannels(0, 1, apm_->kBadNumberChannelsError);
TestChangingReverseChannels(0, apm_->kBadNumberChannelsError);
- for (int i = 1; i < 4; ++i) {
- for (int j = 0; j < 1; ++j) {
+ for (size_t i = 1; i < 4; ++i) {
+ for (size_t j = 0; j < 1; ++j) {
// Output channels much be one or match input channels.
if (j == 1 || i == j) {
TestChangingForwardChannels(i, j, kNoErr);
@@ -880,7 +880,7 @@ TEST_F(ApmTest, Channels) {
EXPECT_EQ(i, apm_->num_input_channels());
EXPECT_EQ(j, apm_->num_output_channels());
// The number of reverse channels used for processing to is always 1.
- EXPECT_EQ(1, apm_->num_reverse_channels());
+ EXPECT_EQ(1u, apm_->num_reverse_channels());
} else {
TestChangingForwardChannels(i, j,
AudioProcessing::kBadNumberChannelsError);
@@ -1294,8 +1294,8 @@ TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) {
const int kSampleRateHz = 16000;
const size_t kSamplesPerChannel =
static_cast<size_t>(AudioProcessing::kChunkSizeMs * kSampleRateHz / 1000);
- const int kNumInputChannels = 2;
- const int kNumOutputChannels = 1;
+ const size_t kNumInputChannels = 2;
+ const size_t kNumOutputChannels = 1;
const size_t kNumChunks = 700;
const float kScaleFactor = 0.25f;
Config config;
@@ -1328,7 +1328,7 @@ TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) {
int_data.get(),
float_data.get(),
&src_buf));
- for (int j = 0; j < kNumInputChannels; ++j) {
+ for (size_t j = 0; j < kNumInputChannels; ++j) {
for (size_t k = 0; k < kSamplesPerChannel; ++k) {
src_buf.channels()[j][k] *= kScaleFactor;
}
@@ -1353,7 +1353,7 @@ TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) {
int_data.get(),
float_data.get(),
&src_buf));
- for (int j = 0; j < kNumInputChannels; ++j) {
+ for (size_t j = 0; j < kNumInputChannels; ++j) {
for (size_t k = 0; k < kSamplesPerChannel; ++k) {
src_buf.channels()[j][k] *= kScaleFactor;
}
@@ -1743,7 +1743,8 @@ void ApmTest::ProcessDebugDump(const std::string& in_filename,
const audioproc::ReverseStream msg = event_msg.reverse_stream();
if (msg.channel_size() > 0) {
- ASSERT_EQ(revframe_->num_channels_, msg.channel_size());
+ ASSERT_EQ(revframe_->num_channels_,
+ static_cast<size_t>(msg.channel_size()));
for (int i = 0; i < msg.channel_size(); ++i) {
memcpy(revfloat_cb_->channels()[i],
msg.channel(i).data(),
@@ -1773,7 +1774,8 @@ void ApmTest::ProcessDebugDump(const std::string& in_filename,
}
if (msg.input_channel_size() > 0) {
- ASSERT_EQ(frame_->num_channels_, msg.input_channel_size());
+ ASSERT_EQ(frame_->num_channels_,
+ static_cast<size_t>(msg.input_channel_size()));
for (int i = 0; i < msg.input_channel_size(); ++i) {
memcpy(float_cb_->channels()[i],
msg.input_channel(i).data(),
@@ -1931,9 +1933,12 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
if (test->num_input_channels() != test->num_output_channels())
continue;
- const int num_render_channels = test->num_reverse_channels();
- const int num_input_channels = test->num_input_channels();
- const int num_output_channels = test->num_output_channels();
+ const size_t num_render_channels =
+ static_cast<size_t>(test->num_reverse_channels());
+ const size_t num_input_channels =
+ static_cast<size_t>(test->num_input_channels());
+ const size_t num_output_channels =
+ static_cast<size_t>(test->num_output_channels());
const size_t samples_per_channel = static_cast<size_t>(
test->sample_rate() * AudioProcessing::kChunkSizeMs / 1000);
@@ -1976,7 +1981,7 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
test->sample_rate(),
LayoutFromChannels(num_output_channels),
float_cb_->channels()));
- for (int j = 0; j < num_output_channels; ++j) {
+ for (size_t j = 0; j < num_output_channels; ++j) {
FloatToS16(float_cb_->channels()[j],
samples_per_channel,
output_cb.channels()[j]);
@@ -2009,7 +2014,7 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveSimilarResults) {
0.01);
// Reset in case of downmixing.
- frame_->num_channels_ = test->num_input_channels();
+ frame_->num_channels_ = static_cast<size_t>(test->num_input_channels());
}
rewind(far_file_);
rewind(near_file_);
@@ -2070,9 +2075,9 @@ TEST_F(ApmTest, Process) {
Init(test->sample_rate(),
test->sample_rate(),
test->sample_rate(),
- test->num_input_channels(),
- test->num_output_channels(),
- test->num_reverse_channels(),
+ static_cast<size_t>(test->num_input_channels()),
+ static_cast<size_t>(test->num_output_channels()),
+ static_cast<size_t>(test->num_reverse_channels()),
true);
int frame_count = 0;
@@ -2097,7 +2102,8 @@ TEST_F(ApmTest, Process) {
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
// Ensure the frame was downmixed properly.
- EXPECT_EQ(test->num_output_channels(), frame_->num_channels_);
+ EXPECT_EQ(static_cast<size_t>(test->num_output_channels()),
+ frame_->num_channels_);
max_output_average += MaxAudioFrame(*frame_);
@@ -2127,7 +2133,7 @@ TEST_F(ApmTest, Process) {
ASSERT_EQ(frame_size, write_count);
// Reset in case of downmixing.
- frame_->num_channels_ = test->num_input_channels();
+ frame_->num_channels_ = static_cast<size_t>(test->num_input_channels());
frame_count++;
}
max_output_average /= frame_count;
@@ -2351,7 +2357,7 @@ class AudioProcessingTest
static void SetUpTestCase() {
// Create all needed output reference files.
const int kNativeRates[] = {8000, 16000, 32000, 48000};
- const int kNumChannels[] = {1, 2};
+ const size_t kNumChannels[] = {1, 2};
for (size_t i = 0; i < arraysize(kNativeRates); ++i) {
for (size_t j = 0; j < arraysize(kNumChannels); ++j) {
for (size_t k = 0; k < arraysize(kNumChannels); ++k) {
@@ -2375,10 +2381,10 @@ class AudioProcessingTest
int output_rate,
int reverse_input_rate,
int reverse_output_rate,
- int num_input_channels,
- int num_output_channels,
- int num_reverse_input_channels,
- int num_reverse_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
+ size_t num_reverse_input_channels,
+ size_t num_reverse_output_channels,
std::string output_file_prefix) {
Config config;
config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
« webrtc/modules/audio_device/include/audio_device_defines.h ('K') | « webrtc/modules/audio_processing/test/audio_file_processor.cc ('k') | webrtc/modules/audio_processing/test/audioproc_float.cc » ('j') | webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h » ('J')

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