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/*
* Copyright (c) 2012 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 "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/voice_engine/utility.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
namespace webrtc {
namespace voe {
namespace {
class UtilityTest : public ::testing::Test {
protected:
UtilityTest() {
src_frame_.sample_rate_hz_ = 16000;
src_frame_.samples_per_channel_ = src_frame_.sample_rate_hz_ / 100;
src_frame_.num_channels_ = 1;
dst_frame_.CopyFrom(src_frame_);
golden_frame_.CopyFrom(src_frame_);
}
void RunResampleTest(int src_channels,
int src_sample_rate_hz,
int dst_channels,
int dst_sample_rate_hz);
PushResampler<int16_t> resampler_;
AudioFrame src_frame_;
AudioFrame dst_frame_;
AudioFrame golden_frame_;
};
// Sets the signal value to increase by |data| with every sample. Floats are
// used so non-integer values result in rounding error, but not an accumulating
// error.
void SetMonoFrame(AudioFrame* frame, float data, int sample_rate_hz) {
memset(frame->data_, 0, sizeof(frame->data_));
frame->num_channels_ = 1;
frame->sample_rate_hz_ = sample_rate_hz;
frame->samples_per_channel_ = sample_rate_hz / 100;
for (size_t i = 0; i < frame->samples_per_channel_; i++) {
frame->data_[i] = static_cast<int16_t>(data * i);
}
}
// Keep the existing sample rate.
void SetMonoFrame(AudioFrame* frame, float data) {
SetMonoFrame(frame, data, frame->sample_rate_hz_);
}
// Sets the signal value to increase by |left| and |right| with every sample in
// each channel respectively.
void SetStereoFrame(AudioFrame* frame, float left, float right,
int sample_rate_hz) {
memset(frame->data_, 0, sizeof(frame->data_));
frame->num_channels_ = 2;
frame->sample_rate_hz_ = sample_rate_hz;
frame->samples_per_channel_ = sample_rate_hz / 100;
for (size_t i = 0; i < frame->samples_per_channel_; i++) {
frame->data_[i * 2] = static_cast<int16_t>(left * i);
frame->data_[i * 2 + 1] = static_cast<int16_t>(right * i);
}
}
// Keep the existing sample rate.
void SetStereoFrame(AudioFrame* frame, float left, float right) {
SetStereoFrame(frame, left, right, frame->sample_rate_hz_);
}
void VerifyParams(const AudioFrame& ref_frame, const AudioFrame& test_frame) {
EXPECT_EQ(ref_frame.num_channels_, test_frame.num_channels_);
EXPECT_EQ(ref_frame.samples_per_channel_, test_frame.samples_per_channel_);
EXPECT_EQ(ref_frame.sample_rate_hz_, test_frame.sample_rate_hz_);
}
// Computes the best SNR based on the error between |ref_frame| and
// |test_frame|. It allows for up to a |max_delay| in samples between the
// signals to compensate for the resampling delay.
float ComputeSNR(const AudioFrame& ref_frame, const AudioFrame& test_frame,
size_t max_delay) {
VerifyParams(ref_frame, test_frame);
float best_snr = 0;
size_t best_delay = 0;
for (size_t delay = 0; delay <= max_delay; delay++) {
float mse = 0;
float variance = 0;
for (size_t i = 0; i < ref_frame.samples_per_channel_ *
ref_frame.num_channels_ - delay; i++) {
int error = ref_frame.data_[i] - test_frame.data_[i + delay];
mse += error * error;
variance += ref_frame.data_[i] * ref_frame.data_[i];
}
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=%" PRIuS "\n", best_snr, best_delay);
return best_snr;
}
void VerifyFramesAreEqual(const AudioFrame& ref_frame,
const AudioFrame& test_frame) {
VerifyParams(ref_frame, test_frame);
for (size_t i = 0;
i < ref_frame.samples_per_channel_ * ref_frame.num_channels_; i++) {
EXPECT_EQ(ref_frame.data_[i], test_frame.data_[i]);
}
}
void UtilityTest::RunResampleTest(int src_channels,
int src_sample_rate_hz,
int dst_channels,
int dst_sample_rate_hz) {
PushResampler<int16_t> resampler; // Create a new one with every test.
const int16_t kSrcLeft = 30; // Shouldn't overflow for any used sample rate.
const int16_t kSrcRight = 15;
const float resampling_factor = (1.0 * 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;
if (src_channels == 1)
SetMonoFrame(&src_frame_, kSrcLeft, src_sample_rate_hz);
else
SetStereoFrame(&src_frame_, kSrcLeft, kSrcRight, src_sample_rate_hz);
if (dst_channels == 1) {
SetMonoFrame(&dst_frame_, 0, dst_sample_rate_hz);
if (src_channels == 1)
SetMonoFrame(&golden_frame_, dst_left, dst_sample_rate_hz);
else
SetMonoFrame(&golden_frame_, dst_mono, dst_sample_rate_hz);
} else {
SetStereoFrame(&dst_frame_, 0, 0, dst_sample_rate_hz);
if (src_channels == 1)
SetStereoFrame(&golden_frame_, dst_left, dst_left, dst_sample_rate_hz);
else
SetStereoFrame(&golden_frame_, dst_left, dst_right, dst_sample_rate_hz);
}
// The sinc resampler has a known delay, which we compute here. Multiplying by
// two gives us a crude maximum for any resampling, as the old resampler
// typically (but not always) has lower delay.
static const size_t kInputKernelDelaySamples = 16;
const size_t max_delay = static_cast<size_t>(
static_cast<double>(dst_sample_rate_hz) / src_sample_rate_hz *
kInputKernelDelaySamples * dst_channels * 2);
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);
RemixAndResample(src_frame_, &resampler, &dst_frame_);
if (src_sample_rate_hz == 96000 && dst_sample_rate_hz == 8000) {
// The sinc resampler gives poor SNR at this extreme conversion, but we
// expect to see this rarely in practice.
EXPECT_GT(ComputeSNR(golden_frame_, dst_frame_, max_delay), 14.0f);
} else {
EXPECT_GT(ComputeSNR(golden_frame_, dst_frame_, max_delay), 46.0f);
}
}
TEST_F(UtilityTest, RemixAndResampleCopyFrameSucceeds) {
// Stereo -> stereo.
SetStereoFrame(&src_frame_, 10, 10);
SetStereoFrame(&dst_frame_, 0, 0);
RemixAndResample(src_frame_, &resampler_, &dst_frame_);
VerifyFramesAreEqual(src_frame_, dst_frame_);
// Mono -> mono.
SetMonoFrame(&src_frame_, 20);
SetMonoFrame(&dst_frame_, 0);
RemixAndResample(src_frame_, &resampler_, &dst_frame_);
VerifyFramesAreEqual(src_frame_, dst_frame_);
}
TEST_F(UtilityTest, RemixAndResampleMixingOnlySucceeds) {
// Stereo -> mono.
SetStereoFrame(&dst_frame_, 0, 0);
SetMonoFrame(&src_frame_, 10);
SetStereoFrame(&golden_frame_, 10, 10);
RemixAndResample(src_frame_, &resampler_, &dst_frame_);
VerifyFramesAreEqual(dst_frame_, golden_frame_);
// Mono -> stereo.
SetMonoFrame(&dst_frame_, 0);
SetStereoFrame(&src_frame_, 10, 20);
SetMonoFrame(&golden_frame_, 15);
RemixAndResample(src_frame_, &resampler_, &dst_frame_);
VerifyFramesAreEqual(golden_frame_, dst_frame_);
}
TEST_F(UtilityTest, RemixAndResampleSucceeds) {
const int kSampleRates[] = {8000, 16000, 32000, 44100, 48000, 96000};
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++) {
RunResampleTest(kChannels[src_channel], kSampleRates[src_rate],
kChannels[dst_channel], kSampleRates[dst_rate]);
}
}
}
}
}
} // namespace
} // namespace voe
} // namespace webrtc
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