blob: 435f928beac86429ed6060796cc8f0f2bc832876 [file] [log] [blame]
/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <math.h>
#include <algorithm>
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_audio/audio_converter.h"
#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
typedef 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 sb(new ChannelBuffer<float>(frames, num_channels));
for (int i = 0; i < num_channels; ++i)
for (int j = 0; j < frames; ++j)
sb->channel(i)[j] = data[i] * j;
return sb;
}
void VerifyParams(const ChannelBuffer<float>& ref,
const ChannelBuffer<float>& test) {
EXPECT_EQ(ref.num_channels(), test.num_channels());
EXPECT_EQ(ref.samples_per_channel(), test.samples_per_channel());
}
// Computes the best SNR based on the error between |ref_frame| and
// |test_frame|. It searches around |expected_delay| in samples between the
// signals to compensate for the resampling delay.
float ComputeSNR(const ChannelBuffer<float>& ref,
const ChannelBuffer<float>& test,
int expected_delay) {
VerifyParams(ref, test);
float best_snr = 0;
int best_delay = 0;
// Search within one sample of the expected delay.
for (int delay = std::max(expected_delay - 1, 0);
delay <= std::min(expected_delay + 1, ref.samples_per_channel());
++delay) {
float mse = 0;
float variance = 0;
float mean = 0;
for (int i = 0; i < ref.num_channels(); ++i) {
for (int j = 0; j < ref.samples_per_channel() - delay; ++j) {
float error = ref.channel(i)[j] - test.channel(i)[j + delay];
mse += error * error;
variance += ref.channel(i)[j] * ref.channel(i)[j];
mean += ref.channel(i)[j];
}
}
const int length = ref.num_channels() * (ref.samples_per_channel() - delay);
mse /= length;
variance /= length;
mean /= length;
variance -= mean * mean;
float snr = 100; // We assign 100 dB to the zero-error case.
if (mse > 0)
snr = 10 * log10(variance / mse);
if (snr > best_snr) {
best_snr = snr;
best_delay = delay;
}
}
printf("SNR=%.1f dB at delay=%d\n", best_snr, best_delay);
return best_snr;
}
// 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,
int src_sample_rate_hz,
int dst_channels,
int dst_sample_rate_hz) {
const float kSrcLeft = 0.0002f;
const float kSrcRight = 0.0001f;
const float resampling_factor = (1.f * src_sample_rate_hz) /
dst_sample_rate_hz;
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;
std::vector<float> src_data(1, kSrcLeft);
if (src_channels == 2)
src_data.push_back(kSrcRight);
ScopedBuffer src_buffer = CreateBuffer(src_data, src_frames);
std::vector<float> dst_data(1, 0);
std::vector<float> ref_data;
if (dst_channels == 1) {
if (src_channels == 1)
ref_data.push_back(dst_left);
else
ref_data.push_back(dst_mono);
} else {
dst_data.push_back(0);
ref_data.push_back(dst_left);
if (src_channels == 1)
ref_data.push_back(dst_left);
else
ref_data.push_back(dst_right);
}
ScopedBuffer dst_buffer = CreateBuffer(dst_data, dst_frames);
ScopedBuffer ref_buffer = CreateBuffer(ref_data, dst_frames);
// The sinc resampler has a known delay, which we compute here.
const int delay_frames = src_sample_rate_hz == dst_sample_rate_hz ? 0 :
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);
AudioConverter converter(src_channels, src_frames, dst_channels, dst_frames);
converter.Convert(src_buffer->channels(), src_channels, src_frames,
dst_channels, dst_frames, dst_buffer->channels());
EXPECT_LT(43.f,
ComputeSNR(*ref_buffer.get(), *dst_buffer.get(), delay_frames));
}
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) {
RunAudioConverterTest(kChannels[src_channel], kSampleRates[src_rate],
kChannels[dst_channel], kSampleRates[dst_rate]);
}
}
}
}
}
} // namespace webrtc