diff options
Diffstat (limited to 'webrtc/common_audio/audio_converter_unittest.cc')
| -rw-r--r-- | webrtc/common_audio/audio_converter_unittest.cc | 38 |
1 files changed, 20 insertions, 18 deletions
diff --git a/webrtc/common_audio/audio_converter_unittest.cc b/webrtc/common_audio/audio_converter_unittest.cc index c85b96e285..dace0bdccf 100644 --- a/webrtc/common_audio/audio_converter_unittest.cc +++ b/webrtc/common_audio/audio_converter_unittest.cc @@ -13,6 +13,7 @@ #include <vector> #include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/base/arraysize.h" #include "webrtc/base/format_macros.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/common_audio/audio_converter.h" @@ -24,11 +25,11 @@ namespace webrtc { typedef rtc::scoped_ptr<ChannelBuffer<float>> ScopedBuffer; // Sets the signal value to increase by |data| with every sample. -ScopedBuffer CreateBuffer(const std::vector<float>& data, int frames) { - const int num_channels = static_cast<int>(data.size()); +ScopedBuffer CreateBuffer(const std::vector<float>& data, size_t frames) { + const size_t num_channels = data.size(); ScopedBuffer sb(new ChannelBuffer<float>(frames, num_channels)); - for (int i = 0; i < num_channels; ++i) - for (int j = 0; j < frames; ++j) + for (size_t i = 0; i < num_channels; ++i) + for (size_t j = 0; j < frames; ++j) sb->channels()[i][j] = data[i] * j; return sb; } @@ -56,7 +57,7 @@ float ComputeSNR(const ChannelBuffer<float>& ref, float mse = 0; float variance = 0; float mean = 0; - for (int i = 0; i < ref.num_channels(); ++i) { + for (size_t i = 0; i < ref.num_channels(); ++i) { for (size_t j = 0; j < ref.num_frames() - delay; ++j) { float error = ref.channels()[i][j] - test.channels()[i][j + delay]; mse += error * error; @@ -85,9 +86,9 @@ float ComputeSNR(const ChannelBuffer<float>& ref, // Sets the source to a linearly increasing signal for which we can easily // generate a reference. Runs the AudioConverter and ensures the output has // sufficiently high SNR relative to the reference. -void RunAudioConverterTest(int src_channels, +void RunAudioConverterTest(size_t src_channels, int src_sample_rate_hz, - int dst_channels, + size_t dst_channels, int dst_sample_rate_hz) { const float kSrcLeft = 0.0002f; const float kSrcRight = 0.0001f; @@ -96,8 +97,8 @@ void RunAudioConverterTest(int src_channels, const float dst_left = resampling_factor * kSrcLeft; const float dst_right = resampling_factor * kSrcRight; const float dst_mono = (dst_left + dst_right) / 2; - const int src_frames = src_sample_rate_hz / 100; - const int dst_frames = dst_sample_rate_hz / 100; + const size_t src_frames = static_cast<size_t>(src_sample_rate_hz / 100); + const size_t dst_frames = static_cast<size_t>(dst_sample_rate_hz / 100); std::vector<float> src_data(1, kSrcLeft); if (src_channels == 2) @@ -127,8 +128,9 @@ void RunAudioConverterTest(int src_channels, static_cast<size_t>( PushSincResampler::AlgorithmicDelaySeconds(src_sample_rate_hz) * dst_sample_rate_hz); - printf("(%d, %d Hz) -> (%d, %d Hz) ", // SNR reported on the same line later. - src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz); + // SNR reported on the same line later. + printf("(%" PRIuS ", %d Hz) -> (%" PRIuS ", %d Hz) ", + src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz); rtc::scoped_ptr<AudioConverter> converter = AudioConverter::Create( src_channels, src_frames, dst_channels, dst_frames); @@ -141,13 +143,13 @@ void RunAudioConverterTest(int src_channels, TEST(AudioConverterTest, ConversionsPassSNRThreshold) { const int kSampleRates[] = {8000, 16000, 32000, 44100, 48000}; - const int kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates); - const int kChannels[] = {1, 2}; - const int kChannelsSize = sizeof(kChannels) / sizeof(*kChannels); - for (int src_rate = 0; src_rate < kSampleRatesSize; ++src_rate) { - for (int dst_rate = 0; dst_rate < kSampleRatesSize; ++dst_rate) { - for (int src_channel = 0; src_channel < kChannelsSize; ++src_channel) { - for (int dst_channel = 0; dst_channel < kChannelsSize; ++dst_channel) { + const size_t kChannels[] = {1, 2}; + for (size_t src_rate = 0; src_rate < arraysize(kSampleRates); ++src_rate) { + for (size_t dst_rate = 0; dst_rate < arraysize(kSampleRates); ++dst_rate) { + for (size_t src_channel = 0; src_channel < arraysize(kChannels); + ++src_channel) { + for (size_t dst_channel = 0; dst_channel < arraysize(kChannels); + ++dst_channel) { RunAudioConverterTest(kChannels[src_channel], kSampleRates[src_rate], kChannels[dst_channel], kSampleRates[dst_rate]); } |